From nshamshiva at gmail.com  Sat May  2 16:56:33 2009
From: nshamshiva at gmail.com (Shiva Shankar N)
Date: Sat, 2 May 2009 20:56:33 -0300
Subject: [Live-devel] RTSP stops after 5 minutes
Message-ID: <4f96b010905021656v195b7086q2d4a94bdabd40506@mail.gmail.com>

Hi All,

 I am using the openRTSP client and it stops working after 5 minutes. And i
download this version on March 21 2009. Is there any bug in the latest
version. I am having even the old version which is working fine.

Thanks
shiv
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From finlayson at live555.com  Sun May  3 01:23:52 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Sun, 3 May 2009 02:23:52 -0600
Subject: [Live-devel] RTSP stops after 5 minutes
In-Reply-To: <4f96b010905021656v195b7086q2d4a94bdabd40506@mail.gmail.com>
References: <4f96b010905021656v195b7086q2d4a94bdabd40506@mail.gmail.com>
Message-ID: <f06240801c62304d2bbd8@[66.80.62.44]>

>  I am using the openRTSP client and it stops working after 5 minutes.

What do you mean "stops working"?

>  And i download this version on March 21 2009.

The latest version of the code is dated April 20th, 2009.  No support 
is given for earlier versions.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From nshamshiva at gmail.com  Sun May  3 08:35:52 2009
From: nshamshiva at gmail.com (Shiva Shankar N)
Date: Sun, 3 May 2009 12:35:52 -0300
Subject: [Live-devel] RTSP stops after 5 minutes
In-Reply-To: <f06240801c62304d2bbd8@66.80.62.44>
References: <4f96b010905021656v195b7086q2d4a94bdabd40506@mail.gmail.com>
	<f06240801c62304d2bbd8@66.80.62.44>
Message-ID: <4f96b010905030835h5024da6fm8c0b2aea0b660329@mail.gmail.com>

Hi Ross,

What do you mean "stops working"?
The application stop or close automatically.. to be exactly after 5 minutes
40 sec

I want to play this application continuously ..  and this the setting i am
using

openrtsp -b 500000 -v -V -w 704 -h 480  -4 -t -c
rtsp://128.xxx.xx.xxx/mpeg4/media.amp >stream.m4v

thanks
shiv



On Sun, May 3, 2009 at 5:23 AM, Ross Finlayson <finlayson at live555.com>wrote:

>  I am using the openRTSP client and it stops working after 5 minutes.
>>
>
> What do you mean "stops working"?
>
>   And i download this version on March 21 2009.
>>
>
> The latest version of the code is dated April 20th, 2009.  No support is
> given for earlier versions.
> --
>
> Ross Finlayson
> Live Networks, Inc.
> http://www.live555.com/
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
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From nshamshiva at gmail.com  Sun May  3 08:41:16 2009
From: nshamshiva at gmail.com (Shiva Shankar N)
Date: Sun, 3 May 2009 12:41:16 -0300
Subject: [Live-devel] RTSP stops after 5 minutes
In-Reply-To: <4f96b010905030835h5024da6fm8c0b2aea0b660329@mail.gmail.com>
References: <4f96b010905021656v195b7086q2d4a94bdabd40506@mail.gmail.com>
	<f06240801c62304d2bbd8@66.80.62.44>
	<4f96b010905030835h5024da6fm8c0b2aea0b660329@mail.gmail.com>
Message-ID: <4f96b010905030841y1b4a5938s8628bbd9f3067a63@mail.gmail.com>

Hi Ross,

 I tried with latest version. I download it today and i checked it. Even the
latest version is stopping after 5 minutes.

openrtsp -b 500000 -v -V -w 704 -h 480  -4 -t -c
rtsp://128.xxx.xx.xxx/mpeg4/media.amp >stream.m4v

thanks
shiv



On Sun, May 3, 2009 at 12:35 PM, Shiva Shankar N <nshamshiva at gmail.com>wrote:

> Hi Ross,
>
> What do you mean "stops working"?
> The application stop or close automatically.. to be exactly after 5 minutes
> 40 sec
>
> I want to play this application continuously ..  and this the setting i am
> using
>
> openrtsp -b 500000 -v -V -w 704 -h 480  -4 -t -c
> rtsp://128.xxx.xx.xxx/mpeg4/media.amp >stream.m4v
>
> thanks
> shiv
>
>
>
>
> On Sun, May 3, 2009 at 5:23 AM, Ross Finlayson <finlayson at live555.com>wrote:
>
>>  I am using the openRTSP client and it stops working after 5 minutes.
>>>
>>
>> What do you mean "stops working"?
>>
>>   And i download this version on March 21 2009.
>>>
>>
>> The latest version of the code is dated April 20th, 2009.  No support is
>> given for earlier versions.
>> --
>>
>> Ross Finlayson
>> Live Networks, Inc.
>> http://www.live555.com/
>> _______________________________________________
>> live-devel mailing list
>> live-devel at lists.live555.com
>> http://lists.live555.com/mailman/listinfo/live-devel
>>
>
>
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From finlayson at live555.com  Sun May  3 14:52:09 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Sun, 3 May 2009 15:52:09 -0600
Subject: [Live-devel] RTSP stops after 5 minutes
In-Reply-To: <4f96b010905030835h5024da6fm8c0b2aea0b660329@mail.gmail.com>
References: <4f96b010905021656v195b7086q2d4a94bdabd40506@mail.gmail.com>
	<f06240801c62304d2bbd8@66.80.62.44>
	<4f96b010905030835h5024da6fm8c0b2aea0b660329@mail.gmail.com>
Message-ID: <f06240801c623c24a0b11@[66.80.62.44]>

>What do you mean "stops working"?
>The application stop or close automatically.. to be exactly after 5 
>minutes 40 sec
>
>I want to play this application continuously ..  and this the 
>setting i am using
>
>openrtsp -b 500000 -v -V -w 704 -h 480  -4 -t -c 
>rtsp://128.xxx.xx.xxx/mpeg4/media.amp >stream.m4v

Please omit the "-V" option, and send us the diagnostic output that 
"openRTSP" displays (to stderr).  This should help explain what's 
happening.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From nshamshiva at gmail.com  Sun May  3 16:35:48 2009
From: nshamshiva at gmail.com (Shiva Shankar N)
Date: Sun, 3 May 2009 20:35:48 -0300
Subject: [Live-devel] RTSP stops after 5 minutes
In-Reply-To: <f06240801c623c24a0b11@66.80.62.44>
References: <4f96b010905021656v195b7086q2d4a94bdabd40506@mail.gmail.com>
	<f06240801c62304d2bbd8@66.80.62.44>
	<4f96b010905030835h5024da6fm8c0b2aea0b660329@mail.gmail.com>
	<f06240801c623c24a0b11@66.80.62.44>
Message-ID: <4f96b010905031635x33ced9c2tfdb338230874911c@mail.gmail.com>

Hi Ross,

 This is error i am getting. please if you now where exactly it going wrong.
 please let me know. So that it would reduce my debugging time


*********************************************************************************************************************************************
User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)


Received OPTIONS response: RTSP/1.0 200 OK
CSeq: 1
Public: DESCRIBE, GET_PARAMETER, PAUSE, PLAY, SETUP, TEARDOWN


Sending request: DESCRIBE rtsp://128.197.178.101/mpeg4/media.amp RTSP/1.0
CSeq: 2
Accept: application/sdp
User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)


Received DESCRIBE response: RTSP/1.0 200 OK
CSeq: 2
Content-Base: rtsp://128.197.178.101:554/mpeg4/media.amp/
Content-Type: application/sdp
Content-Length: 684


Need to read 684 extra bytes
Read 684 extra bytes: v=0
o=- 1241378493017427 1241378493017434 IN IP4 128.197.178.101
s=Media Presentation
e=NONE
c=IN IP4 0.0.0.0
b=AS:8000
t=0 0
a=control:*
a=range:npt=now-
a=mpeg4-iod:
"data:application/mpeg4-iod;base64,AoDUAE8BAf/1AQOAbwABQFBkYXRhOmFw
cGxpY2F0aW9uL21wZWc0LW9kLWF1O2Jhc2U2NCxBUjBCR3dVZkF4Y0F5U1FBWlFRTklCRUVrK0FBZWhJ
QUFIb1NBQVlCQkE9PQQNAQUABAAAAAAAAAAAAAYJAQAAAAAAAAAAAzoAAkA2ZGF0YTphcHBsaWNhdGlv
bi9tcGVnNC1iaWZzLWF1O2Jhc2U2NCx3QkFTWVFTSVVFVUZQd0E9BBICDQAAAgAAAAAAAAAABQMAAEAG
CQEAAAAAAAAAAA=="
m=video 0 RTP/AVP 96
b=AS:8000
a=control:trackID=1
a=rtpmap:96 MP4V-ES/90000
a=fmtp:96 profile-level-id=245;
config=000001B0F5000001B5090000010000000120008C4
019285820F0A21F;
a=mpeg4-esid:201

Opened URL "rtsp://128.197.178.101/mpeg4/media.amp", returning a SDP
description
:
v=0
o=- 1241378493017427 1241378493017434 IN IP4 128.197.178.101
s=Media Presentation
e=NONE
c=IN IP4 0.0.0.0
b=AS:8000
t=0 0
a=control:*
a=range:npt=now-
a=mpeg4-iod:
"data:application/mpeg4-iod;base64,AoDUAE8BAf/1AQOAbwABQFBkYXRhOmFw
cGxpY2F0aW9uL21wZWc0LW9kLWF1O2Jhc2U2NCxBUjBCR3dVZkF4Y0F5U1FBWlFRTklCRUVrK0FBZWhJ
QUFIb1NBQVlCQkE9PQQNAQUABAAAAAAAAAAAAAYJAQAAAAAAAAAAAzoAAkA2ZGF0YTphcHBsaWNhdGlv
bi9tcGVnNC1iaWZzLWF1O2Jhc2U2NCx3QkFTWVFTSVVFVUZQd0E9BBICDQAAAgAAAAAAAAAABQMAAEAG
CQEAAAAAAAAAAA=="
m=video 0 RTP/AVP 96
b=AS:8000
a=control:trackID=1
a=rtpmap:96 MP4V-ES/90000
a=fmtp:96 profile-level-id=245;
config=000001B0F5000001B5090000010000000120008C4
019285820F0A21F;
a=mpeg4-esid:201

Created receiver for "video/MP4V-ES" subsession (client ports 51302-51303)
Sending request: SETUP rtsp://128.197.178.101:554/mpeg4/media.amp/trackID=1RTSP
/1.0
CSeq: 3
Transport: RTP/AVP/TCP;unicast;interleaved=0-1
User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)


Received SETUP response: RTSP/1.0 200 OK
CSeq: 3
Session: 1098178773;timeout=60
Transport: RTP/AVP/TCP;unicast;mode=play;interleaved=116-117


Setup "video/MP4V-ES" subsession (client ports 51302-51303)
Sending request: PLAY rtsp://128.197.178.101:554/mpeg4/media.amp/ RTSP/1.0
CSeq: 4
Session: 1098178773
Range: npt=0.000-
User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)


Received PLAY response: RTSP/1.0 200 OK
CSeq: 4
Session: 1098178773
Range: npt=now-
RTP-Info: url=trackID=1;seq=39372;rtptime=2047243131


Started playing session
Receiving streamed data...
Received RTCP "BYE" on "video/MP4V-ES" subsession (after 313 seconds)
Sending request: PLAY rtsp://128.197.178.101:554/mpeg4/media.amp/ RTSP/1.0
CSeq: 5
Session: 1098178773
Range: npt=0.000-
User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)


Failed to start playing session: Failed to read response: No error
Sending request: TEARDOWN rtsp://128.197.178.101:554/mpeg4/media.amp/RTSP/1.0
CSeq: 6
Session: 1098178773
User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)

Thanks
shiv


On Sun, May 3, 2009 at 6:52 PM, Ross Finlayson <finlayson at live555.com>wrote:

> What do you mean "stops working"?
>> The application stop or close automatically.. to be exactly after 5
>> minutes 40 sec
>>
>> I want to play this application continuously ..  and this the setting i am
>> using
>>
>> openrtsp -b 500000 -v -V -w 704 -h 480  -4 -t -c
>> rtsp://128.xxx.xx.xxx/mpeg4/media.amp >stream.m4v
>>
>
> Please omit the "-V" option, and send us the diagnostic output that
> "openRTSP" displays (to stderr).  This should help explain what's happening.
>
> --
>
> Ross Finlayson
> Live Networks, Inc.
> http://www.live555.com/
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
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From nshamshiva at gmail.com  Sun May  3 17:47:14 2009
From: nshamshiva at gmail.com (Shiva Shankar N)
Date: Sun, 3 May 2009 21:47:14 -0300
Subject: [Live-devel] RTSP stops after 5 minutes
In-Reply-To: <4f96b010905031635x33ced9c2tfdb338230874911c@mail.gmail.com>
References: <4f96b010905021656v195b7086q2d4a94bdabd40506@mail.gmail.com>
	<f06240801c62304d2bbd8@66.80.62.44>
	<4f96b010905030835h5024da6fm8c0b2aea0b660329@mail.gmail.com>
	<f06240801c623c24a0b11@66.80.62.44>
	<4f96b010905031635x33ced9c2tfdb338230874911c@mail.gmail.com>
Message-ID: <4f96b010905031747m53fd93f7x976b46a218dde800@mail.gmail.com>

Hi Ross,

 It is working fine if i tunnel the data thru  UDP. I.e., removing -t option
it is working fine.

thanks
shiv


On Sun, May 3, 2009 at 8:35 PM, Shiva Shankar N <nshamshiva at gmail.com>wrote:

> Hi Ross,
>
>  This is error i am getting. please if you now where exactly it going
> wrong.
>  please let me know. So that it would reduce my debugging time
>
>
>
> *********************************************************************************************************************************************
> User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)
>
>
> Received OPTIONS response: RTSP/1.0 200 OK
> CSeq: 1
> Public: DESCRIBE, GET_PARAMETER, PAUSE, PLAY, SETUP, TEARDOWN
>
>
> Sending request: DESCRIBE rtsp://128.197.178.101/mpeg4/media.amp RTSP/1.0
> CSeq: 2
> Accept: application/sdp
> User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)
>
>
> Received DESCRIBE response: RTSP/1.0 200 OK
> CSeq: 2
> Content-Base: rtsp://128.197.178.101:554/mpeg4/media.amp/
> Content-Type: application/sdp
> Content-Length: 684
>
>
> Need to read 684 extra bytes
> Read 684 extra bytes: v=0
> o=- 1241378493017427 1241378493017434 IN IP4 128.197.178.101
> s=Media Presentation
> e=NONE
> c=IN IP4 0.0.0.0
> b=AS:8000
> t=0 0
> a=control:*
> a=range:npt=now-
> a=mpeg4-iod:
> "data:application/mpeg4-iod;base64,AoDUAE8BAf/1AQOAbwABQFBkYXRhOmFw
>
> cGxpY2F0aW9uL21wZWc0LW9kLWF1O2Jhc2U2NCxBUjBCR3dVZkF4Y0F5U1FBWlFRTklCRUVrK0FBZWhJ
>
> QUFIb1NBQVlCQkE9PQQNAQUABAAAAAAAAAAAAAYJAQAAAAAAAAAAAzoAAkA2ZGF0YTphcHBsaWNhdGlv
>
> bi9tcGVnNC1iaWZzLWF1O2Jhc2U2NCx3QkFTWVFTSVVFVUZQd0E9BBICDQAAAgAAAAAAAAAABQMAAEAG
> CQEAAAAAAAAAAA=="
> m=video 0 RTP/AVP 96
> b=AS:8000
> a=control:trackID=1
> a=rtpmap:96 MP4V-ES/90000
> a=fmtp:96 profile-level-id=245;
> config=000001B0F5000001B5090000010000000120008C4
> 019285820F0A21F;
> a=mpeg4-esid:201
>
> Opened URL "rtsp://128.197.178.101/mpeg4/media.amp", returning a SDP
> description
> :
> v=0
> o=- 1241378493017427 1241378493017434 IN IP4 128.197.178.101
> s=Media Presentation
> e=NONE
> c=IN IP4 0.0.0.0
> b=AS:8000
> t=0 0
> a=control:*
> a=range:npt=now-
> a=mpeg4-iod:
> "data:application/mpeg4-iod;base64,AoDUAE8BAf/1AQOAbwABQFBkYXRhOmFw
>
> cGxpY2F0aW9uL21wZWc0LW9kLWF1O2Jhc2U2NCxBUjBCR3dVZkF4Y0F5U1FBWlFRTklCRUVrK0FBZWhJ
>
> QUFIb1NBQVlCQkE9PQQNAQUABAAAAAAAAAAAAAYJAQAAAAAAAAAAAzoAAkA2ZGF0YTphcHBsaWNhdGlv
>
> bi9tcGVnNC1iaWZzLWF1O2Jhc2U2NCx3QkFTWVFTSVVFVUZQd0E9BBICDQAAAgAAAAAAAAAABQMAAEAG
> CQEAAAAAAAAAAA=="
> m=video 0 RTP/AVP 96
> b=AS:8000
> a=control:trackID=1
> a=rtpmap:96 MP4V-ES/90000
> a=fmtp:96 profile-level-id=245;
> config=000001B0F5000001B5090000010000000120008C4
> 019285820F0A21F;
> a=mpeg4-esid:201
>
> Created receiver for "video/MP4V-ES" subsession (client ports 51302-51303)
> Sending request: SETUP rtsp://
> 128.197.178.101:554/mpeg4/media.amp/trackID=1 RTSP
> /1.0
> CSeq: 3
> Transport: RTP/AVP/TCP;unicast;interleaved=0-1
> User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)
>
>
> Received SETUP response: RTSP/1.0 200 OK
> CSeq: 3
> Session: 1098178773;timeout=60
> Transport: RTP/AVP/TCP;unicast;mode=play;interleaved=116-117
>
>
> Setup "video/MP4V-ES" subsession (client ports 51302-51303)
> Sending request: PLAY rtsp://128.197.178.101:554/mpeg4/media.amp/ RTSP/1.0
> CSeq: 4
> Session: 1098178773
> Range: npt=0.000-
> User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)
>
>
> Received PLAY response: RTSP/1.0 200 OK
> CSeq: 4
> Session: 1098178773
> Range: npt=now-
> RTP-Info: url=trackID=1;seq=39372;rtptime=2047243131
>
>
> Started playing session
> Receiving streamed data...
> Received RTCP "BYE" on "video/MP4V-ES" subsession (after 313 seconds)
> Sending request: PLAY rtsp://128.197.178.101:554/mpeg4/media.amp/ RTSP/1.0
> CSeq: 5
> Session: 1098178773
> Range: npt=0.000-
> User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)
>
>
> Failed to start playing session: Failed to read response: No error
> Sending request: TEARDOWN rtsp://128.197.178.101:554/mpeg4/media.amp/RTSP/1.0
> CSeq: 6
> Session: 1098178773
> User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)
>
> Thanks
> shiv
>
>
>
> On Sun, May 3, 2009 at 6:52 PM, Ross Finlayson <finlayson at live555.com>wrote:
>
>> What do you mean "stops working"?
>>> The application stop or close automatically.. to be exactly after 5
>>> minutes 40 sec
>>>
>>> I want to play this application continuously ..  and this the setting i
>>> am using
>>>
>>> openrtsp -b 500000 -v -V -w 704 -h 480  -4 -t -c
>>> rtsp://128.xxx.xx.xxx/mpeg4/media.amp >stream.m4v
>>>
>>
>> Please omit the "-V" option, and send us the diagnostic output that
>> "openRTSP" displays (to stderr).  This should help explain what's happening.
>>
>> --
>>
>> Ross Finlayson
>> Live Networks, Inc.
>> http://www.live555.com/
>> _______________________________________________
>> live-devel mailing list
>> live-devel at lists.live555.com
>> http://lists.live555.com/mailman/listinfo/live-devel
>>
>
>
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From rmcouat at smartt.com  Sun May  3 20:20:52 2009
From: rmcouat at smartt.com (Ron McOuat)
Date: Sun, 03 May 2009 20:20:52 -0700
Subject: [Live-devel] RTSP stops after 5 minutes
In-Reply-To: <4f96b010905031747m53fd93f7x976b46a218dde800@mail.gmail.com>
References: <4f96b010905021656v195b7086q2d4a94bdabd40506@mail.gmail.com>	<f06240801c62304d2bbd8@66.80.62.44>	<4f96b010905030835h5024da6fm8c0b2aea0b660329@mail.gmail.com>	<f06240801c623c24a0b11@66.80.62.44>	<4f96b010905031635x33ced9c2tfdb338230874911c@mail.gmail.com>
	<4f96b010905031747m53fd93f7x976b46a218dde800@mail.gmail.com>
Message-ID: <49FE5F14.8080907@smartt.com>

I ran into this with Axis cameras, judging from the URL this is what you 
are using. I found the -T option for HTTP tunneling also worked as well 
as leaving off the -t option as you have found.

Shiva Shankar N wrote:
> Hi Ross,
>
>  It is working fine if i tunnel the data thru  UDP. I.e., removing -t 
> option it is working fine.
>
> thanks
> shiv
>
>
> On Sun, May 3, 2009 at 8:35 PM, Shiva Shankar N <nshamshiva at gmail.com 
> <mailto:nshamshiva at gmail.com>> wrote:
>
>     Hi Ross,
>
>      This is error i am getting. please if you now where exactly it
>     going wrong.
>      please let me know. So that it would reduce my debugging time
>
>
>     *********************************************************************************************************************************************
>     User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)
>
>
>     Received OPTIONS response: RTSP/1.0 200 OK
>     CSeq: 1
>     Public: DESCRIBE, GET_PARAMETER, PAUSE, PLAY, SETUP, TEARDOWN
>
>
>     Sending request: DESCRIBE rtsp://128.197.178.101/mpeg4/media.amp
>     <http://128.197.178.101/mpeg4/media.amp> RTSP/1.0
>     CSeq: 2
>     Accept: application/sdp
>     User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)
>
>
>     Received DESCRIBE response: RTSP/1.0 200 OK
>     CSeq: 2
>     Content-Base: rtsp://128.197.178.101:554/mpeg4/media.amp/
>     <http://128.197.178.101:554/mpeg4/media.amp/>
>     Content-Type: application/sdp
>     Content-Length: 684
>
>
>     Need to read 684 extra bytes
>     Read 684 extra bytes: v=0
>     o=- 1241378493017427 1241378493017434 IN IP4 128.197.178.101
>     s=Media Presentation
>     e=NONE
>     c=IN IP4 0.0.0.0
>     b=AS:8000
>     t=0 0
>     a=control:*
>     a=range:npt=now-
>     a=mpeg4-iod:
>     "data:application/mpeg4-iod;base64,AoDUAE8BAf/1AQOAbwABQFBkYXRhOmFw
>     cGxpY2F0aW9uL21wZWc0LW9kLWF1O2Jhc2U2NCxBUjBCR3dVZkF4Y0F5U1FBWlFRTklCRUVrK0FBZWhJ
>     QUFIb1NBQVlCQkE9PQQNAQUABAAAAAAAAAAAAAYJAQAAAAAAAAAAAzoAAkA2ZGF0YTphcHBsaWNhdGlv
>     bi9tcGVnNC1iaWZzLWF1O2Jhc2U2NCx3QkFTWVFTSVVFVUZQd0E9BBICDQAAAgAAAAAAAAAABQMAAEAG
>     CQEAAAAAAAAAAA=="
>     m=video 0 RTP/AVP 96
>     b=AS:8000
>     a=control:trackID=1
>     a=rtpmap:96 MP4V-ES/90000
>     a=fmtp:96 profile-level-id=245;
>     config=000001B0F5000001B5090000010000000120008C4
>     019285820F0A21F;
>     a=mpeg4-esid:201
>
>     Opened URL "rtsp://128.197.178.101/mpeg4/media.amp
>     <http://128.197.178.101/mpeg4/media.amp>", returning a SDP description
>     :
>     v=0
>     o=- 1241378493017427 1241378493017434 IN IP4 128.197.178.101
>     s=Media Presentation
>     e=NONE
>     c=IN IP4 0.0.0.0
>     b=AS:8000
>     t=0 0
>     a=control:*
>     a=range:npt=now-
>     a=mpeg4-iod:
>     "data:application/mpeg4-iod;base64,AoDUAE8BAf/1AQOAbwABQFBkYXRhOmFw
>     cGxpY2F0aW9uL21wZWc0LW9kLWF1O2Jhc2U2NCxBUjBCR3dVZkF4Y0F5U1FBWlFRTklCRUVrK0FBZWhJ
>     QUFIb1NBQVlCQkE9PQQNAQUABAAAAAAAAAAAAAYJAQAAAAAAAAAAAzoAAkA2ZGF0YTphcHBsaWNhdGlv
>     bi9tcGVnNC1iaWZzLWF1O2Jhc2U2NCx3QkFTWVFTSVVFVUZQd0E9BBICDQAAAgAAAAAAAAAABQMAAEAG
>     CQEAAAAAAAAAAA=="
>     m=video 0 RTP/AVP 96
>     b=AS:8000
>     a=control:trackID=1
>     a=rtpmap:96 MP4V-ES/90000
>     a=fmtp:96 profile-level-id=245;
>     config=000001B0F5000001B5090000010000000120008C4
>     019285820F0A21F;
>     a=mpeg4-esid:201
>
>     Created receiver for "video/MP4V-ES" subsession (client ports
>     51302-51303)
>     Sending request: SETUP
>     rtsp://128.197.178.101:554/mpeg4/media.amp/trackID=1
>     <http://128.197.178.101:554/mpeg4/media.amp/trackID=1> RTSP
>     /1.0
>     CSeq: 3
>     Transport: RTP/AVP/TCP;unicast;interleaved=0-1
>     User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)
>
>
>     Received SETUP response: RTSP/1.0 200 OK
>     CSeq: 3
>     Session: 1098178773;timeout=60
>     Transport: RTP/AVP/TCP;unicast;mode=play;interleaved=116-117
>
>
>     Setup "video/MP4V-ES" subsession (client ports 51302-51303)
>     Sending request: PLAY rtsp://128.197.178.101:554/mpeg4/media.amp/
>     <http://128.197.178.101:554/mpeg4/media.amp/> RTSP/1.0
>     CSeq: 4
>     Session: 1098178773
>     Range: npt=0.000-
>     User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)
>
>
>     Received PLAY response: RTSP/1.0 200 OK
>     CSeq: 4
>     Session: 1098178773
>     Range: npt=now-
>     RTP-Info: url=trackID=1;seq=39372;rtptime=2047243131
>
>
>     Started playing session
>     Receiving streamed data...
>     Received RTCP "BYE" on "video/MP4V-ES" subsession (after 313 seconds)
>     Sending request: PLAY rtsp://128.197.178.101:554/mpeg4/media.amp/
>     <http://128.197.178.101:554/mpeg4/media.amp/> RTSP/1.0
>     CSeq: 5
>     Session: 1098178773
>     Range: npt=0.000-
>     User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)
>
>
>     Failed to start playing session: Failed to read response: No error
>     Sending request: TEARDOWN
>     rtsp://128.197.178.101:554/mpeg4/media.amp/
>     <http://128.197.178.101:554/mpeg4/media.amp/> RTSP/1.0
>     CSeq: 6
>     Session: 1098178773
>     User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)
>
>     Thanks
>     shiv
>
>
>
>     On Sun, May 3, 2009 at 6:52 PM, Ross Finlayson
>     <finlayson at live555.com <mailto:finlayson at live555.com>> wrote:
>
>             What do you mean "stops working"?
>             The application stop or close automatically.. to be
>             exactly after 5 minutes 40 sec
>
>             I want to play this application continuously ..  and this
>             the setting i am using
>
>             openrtsp -b 500000 -v -V -w 704 -h 480  -4 -t -c
>             rtsp://128.xxx.xx.xxx/mpeg4/media.amp >stream.m4v
>
>
>         Please omit the "-V" option, and send us the diagnostic output
>         that "openRTSP" displays (to stderr).  This should help
>         explain what's happening.
>
>         -- 
>
>         Ross Finlayson
>         Live Networks, Inc.
>         http://www.live555.com/
>         _______________________________________________
>         live-devel mailing list
>         live-devel at lists.live555.com <mailto:live-devel at lists.live555.com>
>         http://lists.live555.com/mailman/listinfo/live-devel
>
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>   

From ganesh_vijayan at yahoo.com  Mon May  4 04:50:31 2009
From: ganesh_vijayan at yahoo.com (Ganesh V)
Date: Mon, 4 May 2009 04:50:31 -0700 (PDT)
Subject: [Live-devel] RTSP Streaming Playback in QuickTime
Message-ID: <757411.7025.qm@web39506.mail.mud.yahoo.com>

Dear Experts,

I am trying to stream out MPEG4/H.264 streams from my application on a device using livemedia stack. I am able to play the streams without any issues using VLC player and my own player based of Livemedia stack. However, when I try to play the same using a Quicktime player, I observe a "Bad Request" message.

I have updated my Quicktime Player to be the latest and have tried different tips available on Apple developer and other forums. However, I am unable to make any progress. To ensure that I am not introducing any errors, I downloaded Live555 Media Server and streamed out elementary streams using the same. Again, VLC player is able to play the stream without any issues, but Quicktime still fails. For simplicity, I tried only MPEG4 video elementary stream with live555MediaServer.

Any pointers or tips or tricks will be very useful for my progress. Many thanks in advance.

Ganesh


      
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From finlayson at live555.com  Mon May  4 08:03:10 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Mon, 4 May 2009 09:03:10 -0600
Subject: [Live-devel] RTSP Streaming Playback in QuickTime
In-Reply-To: <757411.7025.qm@web39506.mail.mud.yahoo.com>
References: <757411.7025.qm@web39506.mail.mud.yahoo.com>
Message-ID: <f06240804c624b3c3e583@[66.80.62.44]>

>I am trying to stream out MPEG4/H.264 streams from my application on 
>a device using livemedia stack. I am able to play the streams 
>without any issues using VLC player and my own player based of 
>Livemedia stack. However, when I try to play the same using a 
>Quicktime player, I observe a "Bad Request" message.

Turn on debugging output in your RTSP server application, by adding
#define DEBUG 1
to the start of "liveMedia/RTSPServer.cpp", and recompile.  That 
should tell you why the server is rejecting the request.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From v_carvalho_7 at hotmail.com  Mon May  4 08:27:36 2009
From: v_carvalho_7 at hotmail.com (Vitor Carvalho)
Date: Mon, 4 May 2009 15:27:36 +0000
Subject: [Live-devel] RTSP Streaming Playback in QuickTime
In-Reply-To: <f06240804c624b3c3e583@[66.80.62.44]>
References: <757411.7025.qm@web39506.mail.mud.yahoo.com>
	<f06240804c624b3c3e583@[66.80.62.44]>
Message-ID: <BAY134-W9C727A6F765D9205C93D982680@phx.gbl>


Are you using any TI device,like DM355,DM644..??
 
> Date: Mon, 4 May 2009 09:03:10 -0600
> To: live-devel at ns.live555.com
> From: finlayson at live555.com
> Subject: Re: [Live-devel] RTSP Streaming Playback in QuickTime
> 
> >I am trying to stream out MPEG4/H.264 streams from my application on 
> >a device using livemedia stack. I am able to play the streams 
> >without any issues using VLC player and my own player based of 
> >Livemedia stack. However, when I try to play the same using a 
> >Quicktime player, I observe a "Bad Request" message.
> 
> Turn on debugging output in your RTSP server application, by adding
> #define DEBUG 1
> to the start of "liveMedia/RTSPServer.cpp", and recompile. That 
> should tell you why the server is rejecting the request.
> -- 
> 
> Ross Finlayson
> Live Networks, Inc.
> http://www.live555.com/
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel

_________________________________________________________________
Emoticons e Winks super diferentes para o Messenger. Baixe agora, ? gr?tis!
http://specials.br.msn.com/ilovemessenger/pacotes.aspx
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From bartha_adam at yahoo.com  Mon May  4 09:45:40 2009
From: bartha_adam at yahoo.com (Bartha Adam)
Date: Mon, 4 May 2009 09:45:40 -0700 (PDT)
Subject: [Live-devel] multiple multicast stream
Message-ID: <116031.64951.qm@web110304.mail.gq1.yahoo.com>

Hy all!

Maybe I'm a little bit confused...
I am using the testMPEG1or2VideoStreamer.cpp to stream to a fixed multicast address.
It is possible to create multiple streams with the same multicast address and port?

I tried to create 2 streams, but the result seems to contain interleaved frames from the two streams. In this case, what is the importance of the stream name specified in the testMPEG1or2VideoStreamer.cpp?

Regards
Adam





      
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From mrteddy at citromail.hu  Mon May  4 10:56:02 2009
From: mrteddy at citromail.hu (Mr. Teddy)
Date: Mon, 04 May 2009 19:56:02 +0200
Subject: [Live-devel] openRTSP TS receive as an another program input
In-Reply-To: <f06240800c613844da883@[66.80.62.44]>
Message-ID: <20090504175602.16018.qmail@server16.citromail.hu>

            Hello!
 
To pipe openRTSP don't you have a little sample code? I tried _popen() function, but it not the best.
 
Thank you!
 
B.R.: Peter
-- Eredeti ?zenet --
Felad?: Ross Finlayson &lt;finlayson at live555.com&gt;
C?mzett: LIVE555 Streaming Media - development &amp; use &lt;live-devel at ns.live555.com&gt;
M?solat: 
Elk?ldve: 2009.04.21  16:20
T?ma: Re: [Live-devel] openRTSP TS receive as an another program input&gt;And Can I link openRTSP received data buffer to my application input?Yes. Use the "-v" option to cause "openRTSP" to write its output to 'stdout', and then pipe this to your application.-- Ross FinlaysonLive Networks, Inc.http://www.live555.com/_______________________________________________live-devel mailing listlive-devel at lists.live555.comhttp://lists.live555.com/mailman/listinfo/live-devel


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From tony at lava.net  Mon May  4 16:46:50 2009
From: tony at lava.net (Antonio Querubin)
Date: Mon, 4 May 2009 13:46:50 -1000 (HST)
Subject: [Live-devel] multiple multicast stream
In-Reply-To: <116031.64951.qm@web110304.mail.gq1.yahoo.com>
References: <116031.64951.qm@web110304.mail.gq1.yahoo.com>
Message-ID: <alpine.OSX.1.00.0905041344370.561@cust11794.lava.net>

On Mon, 4 May 2009, Bartha Adam wrote:

> Maybe I'm a little bit confused... I am using the 
> testMPEG1or2VideoStreamer.cpp to stream to a fixed multicast address. It 
> is possible to create multiple streams with the same multicast address 
> and port?

If you use the same multicast address and port, you'd need some other way 
of demuxing the streams such as using the source address and SSM.

Antonio Querubin
whois:  AQ7-ARIN
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_______________________________________________
live-devel mailing list
live-devel at lists.live555.com
http://lists.live555.com/mailman/listinfo/live-devel

From ganesh_vijayan at yahoo.com  Mon May  4 17:48:24 2009
From: ganesh_vijayan at yahoo.com (Ganesh V)
Date: Mon, 4 May 2009 17:48:24 -0700 (PDT)
Subject: [Live-devel] RTSP Streaming Playback in QuickTime
In-Reply-To: <BAY134-W9C727A6F765D9205C93D982680@phx.gbl>
References: <757411.7025.qm@web39506.mail.mud.yahoo.com>
	<f06240804c624b3c3e583@[66.80.62.44]>
	<BAY134-W9C727A6F765D9205C93D982680@phx.gbl>
Message-ID: <917199.75049.qm@web39502.mail.mud.yahoo.com>

My device is a DSP chip but not a TI one. However, for my experiment with live555MediaServer, I used my laptop to stream out the data and used VLC on my laptop and other machines to test.



________________________________
From: Vitor Carvalho <v_carvalho_7 at hotmail.com>
To: live-devel at ns.live555.com
Sent: Monday, May 4, 2009 8:57:36 PM
Subject: Re: [Live-devel] RTSP Streaming Playback in QuickTime

 Are you using any TI device,like DM355,DM644..??
 
> Date: Mon, 4 May 2009 09:03:10 -0600
> To: live-devel at ns.live555.com
> From: finlayson at live555.com
> Subject: Re: [Live-devel] RTSP Streaming Playback in QuickTime
> 
> >I am trying to stream out MPEG4/H.264 streams from my application on 
> >a device using livemedia stack. I am able to play the streams 
> >without any issues using VLC player and my own player based of 
> >Livemedia stack. However, when I try to play the same using a 
> >Quicktime player, I observe a "Bad Request" message.
> 
> Turn on debugging output in your RTSP server application, by adding
> #define DEBUG 1
> to the start of "liveMedia/RTSPServer.cpp", and recompile. That 
> should tell you why the server is rejecting the request.
> -- 
> 
> Ross Finlayson
> Live Networks, Inc.
> http://www.live555.com/
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel

________________________________
Quer uma internet mais segura? Baixe agora o novo Internet Explorer 8. ? gr?tis!


      
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From ganesh_vijayan at yahoo.com  Tue May  5 02:04:47 2009
From: ganesh_vijayan at yahoo.com (Ganesh V)
Date: Tue, 5 May 2009 02:04:47 -0700 (PDT)
Subject: [Live-devel] RTSP Streaming Playback in QuickTime
In-Reply-To: <f06240804c624b3c3e583@[66.80.62.44]>
References: <757411.7025.qm@web39506.mail.mud.yahoo.com>
	<f06240804c624b3c3e583@[66.80.62.44]>
Message-ID: <573158.43964.qm@web39502.mail.mud.yahoo.com>

Hello Ross,

Many thanks for your pointer. I enabled debug and retried my experiments with VLC, QT and Windows Media Player. From the logs, I observed the following.

After connection is setup,
- both VLC and QT send DESCRIBE command with bytes and parseRTSPRequestString returns DESCRIBE command name.
- After this stage, both VLC and QT are able to sent SETUP command which is also serviced successfully
- After setup, PLAY is also serviced successfully for both the players. At the end of this stage, RTP info in URL is passed successfully from RTSPServer. 
- After this stage, VLC is able to play the stream without any issues. But, in case of QT, I observed that QT sends "0" bytes and terminates the connection. After the same, QT sends a GET command with HTTP/1.0 to Live555MediaServer. 
- Because the incoming protocol type is HTTP, parseRTSPRequestString() failed and hence, Bad Request Message is observed in QT.

The same issue is observed when I try playing the streaming data using Windows Media Player too. I have attached the log in case of QT player.

Is this a known defect with QT or am I missing some parameter or initialization? Another interesting issue I observed was that one version of QT detected the frame width and height, but was unable to display the decoded data. 

Do I need to invoke or modify any call in case of QT/ Windows Media Player? I assume that the current failure is due to the incoming request type to be HTTP. How do I overcome the same?

Thanks,
Ganesh





________________________________
From: Ross Finlayson <finlayson at live555.com>
To: LIVE555 Streaming Media - development & use <live-devel at ns.live555.com>
Sent: Monday, May 4, 2009 8:33:10 PM
Subject: Re: [Live-devel] RTSP Streaming Playback in QuickTime

> I am trying to stream out MPEG4/H.264 streams from my application on a device using livemedia stack. I am able to play the streams without any issues using VLC player and my own player based of Livemedia stack. However, when I try to play the same using a Quicktime player, I observe a "Bad Request" message.

Turn on debugging output in your RTSP server application, by adding
#define DEBUG 1
to the start of "liveMedia/RTSPServer.cpp", and recompile.  That should tell you why the server is rejecting the request.
-- 
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
_______________________________________________
live-devel mailing list
live-devel at lists.live555.com
http://lists.live555.com/mailman/listinfo/live-devel



      
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From finlayson at live555.com  Tue May  5 06:22:20 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Tue, 5 May 2009 07:22:20 -0600
Subject: [Live-devel] RTSP Streaming Playback in QuickTime
In-Reply-To: <573158.43964.qm@web39502.mail.mud.yahoo.com>
References: <757411.7025.qm@web39506.mail.mud.yahoo.com>
	<f06240804c624b3c3e583@[66.80.62.44]>
	<573158.43964.qm@web39502.mail.mud.yahoo.com>
Message-ID: <f06240800c625ed5db8a4@[66.80.62.44]>

>- Because the incoming protocol type is HTTP, 
>parseRTSPRequestString() failed and hence, Bad Request Message is 
>observed in QT.

Yes, the problem is that your client (QuickTime Player) is attempting 
to use RTSP-over-HTTP, which we do not yet support in our RTSP server 
implementation.

Presumably you have a firewall somewhere between your server and 
client, otherwise it would probably have tried regular RTSP (not over 
HTTP).  If you fix this, QuickTime Player should be able to connect 
to your server.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From finlayson at live555.com  Tue May  5 06:31:35 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Tue, 5 May 2009 07:31:35 -0600
Subject: [Live-devel] openRTSP TS receive as an another program input
In-Reply-To: <20090504175602.16018.qmail@server16.citromail.hu>
References: <20090504175602.16018.qmail@server16.citromail.hu>
Message-ID: <f06240801c625efc348ae@[66.80.62.44]>

>To pipe openRTSP don't you have a little sample code? I tried 
>_popen() function, but it not the best.

No, you don't need to write any code for this; you can do this from 
the command line

	openRTSP -v etc. rtsp://url | your-application-that-reads-from-stdin
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From finlayson at live555.com  Tue May  5 06:36:27 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Tue, 5 May 2009 07:36:27 -0600
Subject: [Live-devel] multiple multicast stream
In-Reply-To: <116031.64951.qm@web110304.mail.gq1.yahoo.com>
References: <116031.64951.qm@web110304.mail.gq1.yahoo.com>
Message-ID: <f06240802c625f0967a00@[66.80.62.44]>

>I am using the testMPEG1or2VideoStreamer.cpp to stream to a fixed 
>multicast address.
>It is possible to create multiple streams with the same multicast 
>address and port?

No.  Multiple streams should each use separate multicast addresses 
and port numbers.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From ganesh_vijayan at yahoo.com  Tue May  5 07:22:48 2009
From: ganesh_vijayan at yahoo.com (Ganesh V)
Date: Tue, 5 May 2009 07:22:48 -0700 (PDT)
Subject: [Live-devel] RTSP Streaming Playback in QuickTime
In-Reply-To: <f06240800c625ed5db8a4@[66.80.62.44]>
References: <757411.7025.qm@web39506.mail.mud.yahoo.com>
	<f06240804c624b3c3e583@[66.80.62.44]>
	<573158.43964.qm@web39502.mail.mud.yahoo.com>
	<f06240800c625ed5db8a4@[66.80.62.44]>
Message-ID: <325478.59598.qm@web39507.mail.mud.yahoo.com>

Thank you, Ross. Windows firewall was indeed the issue. I switched off the windows firewall and was able to play the streams in QT. I will proceed with debugging my application with QT.

Thanks,
Ganesh




________________________________
From: Ross Finlayson <finlayson at live555.com>
To: LIVE555 Streaming Media - development & use <live-devel at ns.live555.com>
Sent: Tuesday, May 5, 2009 6:52:20 PM
Subject: Re: [Live-devel] RTSP Streaming Playback in QuickTime

> - Because the incoming protocol type is HTTP, parseRTSPRequestString() failed and hence, Bad Request Message is observed in QT.

Yes, the problem is that your client (QuickTime Player) is attempting to use RTSP-over-HTTP, which we do not yet support in our RTSP server implementation.

Presumably you have a firewall somewhere between your server and client, otherwise it would probably have tried regular RTSP (not over HTTP).  If you fix this, QuickTime Player should be able to connect to your server.
-- 
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
_______________________________________________
live-devel mailing list
live-devel at lists.live555.com
http://lists.live555.com/mailman/listinfo/live-devel



      
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From nshamshiva at gmail.com  Tue May  5 10:01:31 2009
From: nshamshiva at gmail.com (Shiva Shankar N)
Date: Tue, 5 May 2009 14:01:31 -0300
Subject: [Live-devel] RTSP stops after 5 minutes
In-Reply-To: <4f96b010905032204y6465ac56vfd9ff3832e8afca1@mail.gmail.com>
References: <4f96b010905021656v195b7086q2d4a94bdabd40506@mail.gmail.com>
	<f06240801c62304d2bbd8@66.80.62.44>
	<4f96b010905030835h5024da6fm8c0b2aea0b660329@mail.gmail.com>
	<f06240801c623c24a0b11@66.80.62.44>
	<4f96b010905031635x33ced9c2tfdb338230874911c@mail.gmail.com>
	<4f96b010905031747m53fd93f7x976b46a218dde800@mail.gmail.com>
	<49FE5F14.8080907@smartt.com>
	<4f96b010905032204y6465ac56vfd9ff3832e8afca1@mail.gmail.com>
Message-ID: <4f96b010905051001u2aa3f8bbpc1c677784fe187d5@mail.gmail.com>

Hi

 this is there any way that i would make openrtsp client streaming to work
for more than 5 minutes with TCP tunneling  i.e., with -t option.

thanking you
shiv


On Mon, May 4, 2009 at 2:04 AM, Shiva Shankar N <nshamshiva at gmail.com>wrote:

> Hi Ross,
>
> Thanks for your quick reply.
> Do you have any suggestion for me to fix this bug. I am really in urgent to
> finish off this project. I saw some of your previous posting in 2007 (appox
> april) there was a same problem. And you had release new code with the bug
> fixed.
>
> thanks
> shiv
>
>
>
> On Mon, May 4, 2009 at 12:20 AM, Ron McOuat <rmcouat at smartt.com> wrote:
>
>> I ran into this with Axis cameras, judging from the URL this is what you
>> are using. I found the -T option for HTTP tunneling also worked as well as
>> leaving off the -t option as you have found.
>>
>> Shiva Shankar N wrote:
>>
>>> Hi Ross,
>>>
>>>  It is working fine if i tunnel the data thru  UDP. I.e., removing -t
>>> option it is working fine.
>>>
>>> thanks
>>> shiv
>>>
>>>
>>> On Sun, May 3, 2009 at 8:35 PM, Shiva Shankar N <nshamshiva at gmail.com<mailto:
>>> nshamshiva at gmail.com>> wrote:
>>>
>>>    Hi Ross,
>>>
>>>     This is error i am getting. please if you now where exactly it
>>>    going wrong.
>>>     please let me know. So that it would reduce my debugging time
>>>
>>>
>>>
>>>  *********************************************************************************************************************************************
>>>    User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)
>>>
>>>
>>>    Received OPTIONS response: RTSP/1.0 200 OK
>>>    CSeq: 1
>>>    Public: DESCRIBE, GET_PARAMETER, PAUSE, PLAY, SETUP, TEARDOWN
>>>
>>>
>>>    Sending request: DESCRIBE rtsp://128.197.178.101/mpeg4/media.amp
>>>    <http://128.197.178.101/mpeg4/media.amp> RTSP/1.0
>>>    CSeq: 2
>>>    Accept: application/sdp
>>>    User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)
>>>
>>>
>>>    Received DESCRIBE response: RTSP/1.0 200 OK
>>>    CSeq: 2
>>>    Content-Base: rtsp://128.197.178.101:554/mpeg4/media.amp/
>>>    <http://128.197.178.101:554/mpeg4/media.amp/>
>>>
>>>    Content-Type: application/sdp
>>>    Content-Length: 684
>>>
>>>
>>>    Need to read 684 extra bytes
>>>    Read 684 extra bytes: v=0
>>>    o=- 1241378493017427 1241378493017434 IN IP4 128.197.178.101
>>>    s=Media Presentation
>>>    e=NONE
>>>    c=IN IP4 0.0.0.0
>>>    b=AS:8000
>>>    t=0 0
>>>    a=control:*
>>>    a=range:npt=now-
>>>    a=mpeg4-iod:
>>>    "data:application/mpeg4-iod;base64,AoDUAE8BAf/1AQOAbwABQFBkYXRhOmFw
>>>
>>>  cGxpY2F0aW9uL21wZWc0LW9kLWF1O2Jhc2U2NCxBUjBCR3dVZkF4Y0F5U1FBWlFRTklCRUVrK0FBZWhJ
>>>
>>>  QUFIb1NBQVlCQkE9PQQNAQUABAAAAAAAAAAAAAYJAQAAAAAAAAAAAzoAAkA2ZGF0YTphcHBsaWNhdGlv
>>>
>>>  bi9tcGVnNC1iaWZzLWF1O2Jhc2U2NCx3QkFTWVFTSVVFVUZQd0E9BBICDQAAAgAAAAAAAAAABQMAAEAG
>>>    CQEAAAAAAAAAAA=="
>>>    m=video 0 RTP/AVP 96
>>>    b=AS:8000
>>>    a=control:trackID=1
>>>    a=rtpmap:96 MP4V-ES/90000
>>>    a=fmtp:96 profile-level-id=245;
>>>    config=000001B0F5000001B5090000010000000120008C4
>>>    019285820F0A21F;
>>>    a=mpeg4-esid:201
>>>
>>>    Opened URL "rtsp://128.197.178.101/mpeg4/media.amp
>>>    <http://128.197.178.101/mpeg4/media.amp>", returning a SDP
>>> description
>>>
>>>    :
>>>    v=0
>>>    o=- 1241378493017427 1241378493017434 IN IP4 128.197.178.101
>>>    s=Media Presentation
>>>    e=NONE
>>>    c=IN IP4 0.0.0.0
>>>    b=AS:8000
>>>    t=0 0
>>>    a=control:*
>>>    a=range:npt=now-
>>>    a=mpeg4-iod:
>>>    "data:application/mpeg4-iod;base64,AoDUAE8BAf/1AQOAbwABQFBkYXRhOmFw
>>>
>>>  cGxpY2F0aW9uL21wZWc0LW9kLWF1O2Jhc2U2NCxBUjBCR3dVZkF4Y0F5U1FBWlFRTklCRUVrK0FBZWhJ
>>>
>>>  QUFIb1NBQVlCQkE9PQQNAQUABAAAAAAAAAAAAAYJAQAAAAAAAAAAAzoAAkA2ZGF0YTphcHBsaWNhdGlv
>>>
>>>  bi9tcGVnNC1iaWZzLWF1O2Jhc2U2NCx3QkFTWVFTSVVFVUZQd0E9BBICDQAAAgAAAAAAAAAABQMAAEAG
>>>    CQEAAAAAAAAAAA=="
>>>    m=video 0 RTP/AVP 96
>>>    b=AS:8000
>>>    a=control:trackID=1
>>>    a=rtpmap:96 MP4V-ES/90000
>>>    a=fmtp:96 profile-level-id=245;
>>>    config=000001B0F5000001B5090000010000000120008C4
>>>    019285820F0A21F;
>>>    a=mpeg4-esid:201
>>>
>>>    Created receiver for "video/MP4V-ES" subsession (client ports
>>>    51302-51303)
>>>    Sending request: SETUP
>>>    rtsp://128.197.178.101:554/mpeg4/media.amp/trackID=1
>>>    <http://128.197.178.101:554/mpeg4/media.amp/trackID=1> RTSP
>>>    /1.0
>>>    CSeq: 3
>>>    Transport: RTP/AVP/TCP;unicast;interleaved=0-1
>>>    User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)
>>>
>>>
>>>    Received SETUP response: RTSP/1.0 200 OK
>>>    CSeq: 3
>>>    Session: 1098178773;timeout=60
>>>    Transport: RTP/AVP/TCP;unicast;mode=play;interleaved=116-117
>>>
>>>
>>>    Setup "video/MP4V-ES" subsession (client ports 51302-51303)
>>>    Sending request: PLAY rtsp://128.197.178.101:554/mpeg4/media.amp/
>>>    <http://128.197.178.101:554/mpeg4/media.amp/> RTSP/1.0
>>>    CSeq: 4
>>>    Session: 1098178773
>>>    Range: npt=0.000-
>>>    User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)
>>>
>>>
>>>    Received PLAY response: RTSP/1.0 200 OK
>>>    CSeq: 4
>>>    Session: 1098178773
>>>    Range: npt=now-
>>>    RTP-Info: url=trackID=1;seq=39372;rtptime=2047243131
>>>
>>>
>>>    Started playing session
>>>    Receiving streamed data...
>>>    Received RTCP "BYE" on "video/MP4V-ES" subsession (after 313 seconds)
>>>    Sending request: PLAY rtsp://128.197.178.101:554/mpeg4/media.amp/
>>>    <http://128.197.178.101:554/mpeg4/media.amp/> RTSP/1.0
>>>    CSeq: 5
>>>    Session: 1098178773
>>>    Range: npt=0.000-
>>>    User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)
>>>
>>>
>>>    Failed to start playing session: Failed to read response: No error
>>>    Sending request: TEARDOWN
>>>    rtsp://128.197.178.101:554/mpeg4/media.amp/
>>>    <http://128.197.178.101:554/mpeg4/media.amp/> RTSP/1.0
>>>    CSeq: 6
>>>    Session: 1098178773
>>>    User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)
>>>
>>>    Thanks
>>>    shiv
>>>
>>>
>>>
>>>    On Sun, May 3, 2009 at 6:52 PM, Ross Finlayson
>>>    <finlayson at live555.com <mailto:finlayson at live555.com>> wrote:
>>>
>>>            What do you mean "stops working"?
>>>            The application stop or close automatically.. to be
>>>            exactly after 5 minutes 40 sec
>>>
>>>            I want to play this application continuously ..  and this
>>>            the setting i am using
>>>
>>>            openrtsp -b 500000 -v -V -w 704 -h 480  -4 -t -c
>>>            rtsp://128.xxx.xx.xxx/mpeg4/media.amp >stream.m4v
>>>
>>>
>>>        Please omit the "-V" option, and send us the diagnostic output
>>>        that "openRTSP" displays (to stderr).  This should help
>>>        explain what's happening.
>>>
>>>        --
>>>        Ross Finlayson
>>>        Live Networks, Inc.
>>>        http://www.live555.com/
>>>        _______________________________________________
>>>        live-devel mailing list
>>>        live-devel at lists.live555.com <mailto:live-devel at lists.live555.com
>>> >
>>>        http://lists.live555.com/mailman/listinfo/live-devel
>>>
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> live-devel mailing list
>>> live-devel at lists.live555.com
>>> http://lists.live555.com/mailman/listinfo/live-devel
>>>
>>>
>> _______________________________________________
>> live-devel mailing list
>> live-devel at lists.live555.com
>> http://lists.live555.com/mailman/listinfo/live-devel
>>
>
>
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From finlayson at live555.com  Tue May  5 22:28:01 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Tue, 5 May 2009 23:28:01 -0600
Subject: [Live-devel] RTSP stops after 5 minutes
In-Reply-To: <4f96b010905051001u2aa3f8bbpc1c677784fe187d5@mail.gmail.com>
References: <4f96b010905021656v195b7086q2d4a94bdabd40506@mail.gmail.com>
	<f06240801c62304d2bbd8@66.80.62.44>
	<4f96b010905030835h5024da6fm8c0b2aea0b660329@mail.gmail.com>
	<f06240801c623c24a0b11@66.80.62.44>
	<4f96b010905031635x33ced9c2tfdb338230874911c@mail.gmail.com>
	<4f96b010905031747m53fd93f7x976b46a218dde800@mail.gmail.com>
	<49FE5F14.8080907@smartt.com>
	<4f96b010905032204y6465ac56vfd9ff3832e8afca1@mail.gmail.com>
	<4f96b010905051001u2aa3f8bbpc1c677784fe187d5@mail.gmail.com>
Message-ID: <f06240807c626d03ac06f@[66.80.62.44]>

>  this is there any way that i would make openrtsp client streaming 
>to work for more than 5 minutes with TCP tunneling  i.e., with -t 
>option.

Sorry, but youre going to have to figure out for yourself why this is 
not working for you.  Remember, You Have Complete Source Code.

-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From mrteddy at citromail.hu  Tue May  5 23:03:40 2009
From: mrteddy at citromail.hu (Mr. Teddy)
Date: Wed, 06 May 2009 08:03:40 +0200
Subject: [Live-devel] openRTSP TS receive as an another program input
In-Reply-To: <f06240801c625efc348ae@[66.80.62.44]>
Message-ID: <20090506060340.32012.qmail@server16.citromail.hu>

            Hello,
 
My application is reads TS from file (example video.ts) with this function:
 
FILE* fp = fopen("video.ts", &#8220;rb&#8221;);
 
If I use openRTSP like you wrote, and if I modify my application fopen to this:
 
FILE* fp=freopen("CONIN$", "rb", stdin);
 
In this two case, should I probably get the same result?
Or sould I change the method to read from stdin? You probably have some experience with it.
 
Very thank You, for your help!
 
Best Regards:
Peter
-- Eredeti ?zenet --
Felad?: Ross Finlayson &lt;finlayson at live555.com&gt;
C?mzett: LIVE555 Streaming Media - development &amp; use &lt;live-devel at ns.live555.com&gt;
M?solat: 
Elk?ldve: 2009.05.05  15:51
T?ma: Re: [Live-devel] openRTSP TS receive as an another program input&gt;To pipe openRTSP don't you have a little sample code? I tried &gt;_popen() function, but it not the best.No, you don't need to write any code for this; you can do this from the command lineopenRTSP -v etc. rtsp://url | your-application-that-reads-from-stdin-- Ross FinlaysonLive Networks, Inc.http://www.live555.com/_______________________________________________live-devel mailing listlive-devel at lists.live555.comhttp://lists.live555.com/mailman/listinfo/live-devel


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From rmcouat at smartt.com  Wed May  6 00:50:14 2009
From: rmcouat at smartt.com (Ron McOuat)
Date: Wed, 06 May 2009 00:50:14 -0700
Subject: [Live-devel] RTSP stops after 5 minutes
In-Reply-To: <4f96b010905051001u2aa3f8bbpc1c677784fe187d5@mail.gmail.com>
References: <4f96b010905021656v195b7086q2d4a94bdabd40506@mail.gmail.com>	<f06240801c62304d2bbd8@66.80.62.44>	<4f96b010905030835h5024da6fm8c0b2aea0b660329@mail.gmail.com>	<f06240801c623c24a0b11@66.80.62.44>	<4f96b010905031635x33ced9c2tfdb338230874911c@mail.gmail.com>	<4f96b010905031747m53fd93f7x976b46a218dde800@mail.gmail.com>	<49FE5F14.8080907@smartt.com>	<4f96b010905032204y6465ac56vfd9ff3832e8afca1@mail.gmail.com>
	<4f96b010905051001u2aa3f8bbpc1c677784fe187d5@mail.gmail.com>
Message-ID: <4A014136.10202@smartt.com>

The media.amp ending on the URL says you are using an Axis camera. If 
you replace the -t option with -T 80 you will switch from using regular 
RTSP/RTP by TCP to TCP tunneled on HTTP to port 80 which the Axis camera 
will accept. This did not fail for me after a time period. I compiled 
live555 with the -DDEBUG option added to CFLAGS and followed the trace 
for the sessions quiting with the -t option. I could not get to the 
bottom of it because of time pressure and I had for me a workable 
solution using -T 80. I posted the trace to the list over a year ago but 
as Ross says it requires getting in and tracing it in detail. I also 
found the amount of time before failure varied up to 30-40 minutes in my 
case. In the end I didn't know if I was fighting a problem in the LAN, a 
problem with the Axis camera or a live555 problem. Axis has released 
patches to firmware for some models that mention fixes to the MPEG4 
image transfer mode so you should look at that as well.

Shiva Shankar N wrote:
> Hi
>
>  this is there any way that i would make openrtsp client streaming to 
> work for more than 5 minutes with TCP tunneling  i.e., with -t option.
>
> thanking you
> shiv
>
>
> On Mon, May 4, 2009 at 2:04 AM, Shiva Shankar N <nshamshiva at gmail.com 
> <mailto:nshamshiva at gmail.com>> wrote:
>
>     Hi Ross,
>      
>     Thanks for your quick reply.
>     Do you have any suggestion for me to fix this bug. I am really in
>     urgent to finish off this project. I saw some of your previous
>     posting in 2007 (appox april) there was a same problem. And you
>     had release new code with the bug fixed.
>
>     thanks
>     shiv
>
>
>
>     On Mon, May 4, 2009 at 12:20 AM, Ron McOuat <rmcouat at smartt.com
>     <mailto:rmcouat at smartt.com>> wrote:
>
>         I ran into this with Axis cameras, judging from the URL this
>         is what you are using. I found the -T option for HTTP
>         tunneling also worked as well as leaving off the -t option as
>         you have found.
>
>         Shiva Shankar N wrote:
>
>             Hi Ross,
>
>              It is working fine if i tunnel the data thru  UDP. I.e.,
>             removing -t option it is working fine.
>
>             thanks
>             shiv
>
>
>             On Sun, May 3, 2009 at 8:35 PM, Shiva Shankar N
>             <nshamshiva at gmail.com <mailto:nshamshiva at gmail.com>
>             <mailto:nshamshiva at gmail.com
>             <mailto:nshamshiva at gmail.com>>> wrote:
>
>                Hi Ross,
>
>                 This is error i am getting. please if you now where
>             exactly it
>                going wrong.
>                 please let me know. So that it would reduce my
>             debugging time
>
>
>              
>              *********************************************************************************************************************************************
>                User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)
>
>
>                Received OPTIONS response: RTSP/1.0 200 OK
>                CSeq: 1
>                Public: DESCRIBE, GET_PARAMETER, PAUSE, PLAY, SETUP,
>             TEARDOWN
>
>
>                Sending request: DESCRIBE
>             rtsp://128.197.178.101/mpeg4/media.amp
>             <http://128.197.178.101/mpeg4/media.amp>
>                <http://128.197.178.101/mpeg4/media.amp> RTSP/1.0
>
>                CSeq: 2
>                Accept: application/sdp
>                User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)
>
>
>                Received DESCRIBE response: RTSP/1.0 200 OK
>                CSeq: 2
>                Content-Base:
>             rtsp://128.197.178.101:554/mpeg4/media.amp/
>             <http://128.197.178.101:554/mpeg4/media.amp/>
>                <http://128.197.178.101:554/mpeg4/media.amp/>
>
>                Content-Type: application/sdp
>                Content-Length: 684
>
>
>                Need to read 684 extra bytes
>                Read 684 extra bytes: v=0
>                o=- 1241378493017427 1241378493017434 IN IP4
>             128.197.178.101
>                s=Media Presentation
>                e=NONE
>                c=IN IP4 0.0.0.0
>                b=AS:8000
>                t=0 0
>                a=control:*
>                a=range:npt=now-
>                a=mpeg4-iod:
>              
>              "data:application/mpeg4-iod;base64,AoDUAE8BAf/1AQOAbwABQFBkYXRhOmFw
>              
>              cGxpY2F0aW9uL21wZWc0LW9kLWF1O2Jhc2U2NCxBUjBCR3dVZkF4Y0F5U1FBWlFRTklCRUVrK0FBZWhJ
>              
>              QUFIb1NBQVlCQkE9PQQNAQUABAAAAAAAAAAAAAYJAQAAAAAAAAAAAzoAAkA2ZGF0YTphcHBsaWNhdGlv
>              
>              bi9tcGVnNC1iaWZzLWF1O2Jhc2U2NCx3QkFTWVFTSVVFVUZQd0E9BBICDQAAAgAAAAAAAAAABQMAAEAG
>                CQEAAAAAAAAAAA=="
>                m=video 0 RTP/AVP 96
>                b=AS:8000
>                a=control:trackID=1
>                a=rtpmap:96 MP4V-ES/90000
>                a=fmtp:96 profile-level-id=245;
>                config=000001B0F5000001B5090000010000000120008C4
>                019285820F0A21F;
>                a=mpeg4-esid:201
>
>                Opened URL "rtsp://128.197.178.101/mpeg4/media.amp
>             <http://128.197.178.101/mpeg4/media.amp>
>                <http://128.197.178.101/mpeg4/media.amp>", returning a
>             SDP description
>
>                :
>                v=0
>                o=- 1241378493017427 1241378493017434 IN IP4
>             128.197.178.101
>                s=Media Presentation
>                e=NONE
>                c=IN IP4 0.0.0.0
>                b=AS:8000
>                t=0 0
>                a=control:*
>                a=range:npt=now-
>                a=mpeg4-iod:
>              
>              "data:application/mpeg4-iod;base64,AoDUAE8BAf/1AQOAbwABQFBkYXRhOmFw
>              
>              cGxpY2F0aW9uL21wZWc0LW9kLWF1O2Jhc2U2NCxBUjBCR3dVZkF4Y0F5U1FBWlFRTklCRUVrK0FBZWhJ
>              
>              QUFIb1NBQVlCQkE9PQQNAQUABAAAAAAAAAAAAAYJAQAAAAAAAAAAAzoAAkA2ZGF0YTphcHBsaWNhdGlv
>              
>              bi9tcGVnNC1iaWZzLWF1O2Jhc2U2NCx3QkFTWVFTSVVFVUZQd0E9BBICDQAAAgAAAAAAAAAABQMAAEAG
>                CQEAAAAAAAAAAA=="
>                m=video 0 RTP/AVP 96
>                b=AS:8000
>                a=control:trackID=1
>                a=rtpmap:96 MP4V-ES/90000
>                a=fmtp:96 profile-level-id=245;
>                config=000001B0F5000001B5090000010000000120008C4
>                019285820F0A21F;
>                a=mpeg4-esid:201
>
>                Created receiver for "video/MP4V-ES" subsession (client
>             ports
>                51302-51303)
>                Sending request: SETUP
>                rtsp://128.197.178.101:554/mpeg4/media.amp/trackID=1
>             <http://128.197.178.101:554/mpeg4/media.amp/trackID=1>
>                <http://128.197.178.101:554/mpeg4/media.amp/trackID=1>
>             RTSP
>
>                /1.0
>                CSeq: 3
>                Transport: RTP/AVP/TCP;unicast;interleaved=0-1
>                User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)
>
>
>                Received SETUP response: RTSP/1.0 200 OK
>                CSeq: 3
>                Session: 1098178773;timeout=60
>                Transport:
>             RTP/AVP/TCP;unicast;mode=play;interleaved=116-117
>
>
>                Setup "video/MP4V-ES" subsession (client ports 51302-51303)
>                Sending request: PLAY
>             rtsp://128.197.178.101:554/mpeg4/media.amp/
>             <http://128.197.178.101:554/mpeg4/media.amp/>
>                <http://128.197.178.101:554/mpeg4/media.amp/> RTSP/1.0
>
>                CSeq: 4
>                Session: 1098178773
>                Range: npt=0.000-
>                User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)
>
>
>                Received PLAY response: RTSP/1.0 200 OK
>                CSeq: 4
>                Session: 1098178773
>                Range: npt=now-
>                RTP-Info: url=trackID=1;seq=39372;rtptime=2047243131
>
>
>                Started playing session
>                Receiving streamed data...
>                Received RTCP "BYE" on "video/MP4V-ES" subsession
>             (after 313 seconds)
>                Sending request: PLAY
>             rtsp://128.197.178.101:554/mpeg4/media.amp/
>             <http://128.197.178.101:554/mpeg4/media.amp/>
>                <http://128.197.178.101:554/mpeg4/media.amp/> RTSP/1.0
>
>                CSeq: 5
>                Session: 1098178773
>                Range: npt=0.000-
>                User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)
>
>
>                Failed to start playing session: Failed to read
>             response: No error
>                Sending request: TEARDOWN
>                rtsp://128.197.178.101:554/mpeg4/media.amp/
>             <http://128.197.178.101:554/mpeg4/media.amp/>
>                <http://128.197.178.101:554/mpeg4/media.amp/> RTSP/1.0
>
>                CSeq: 6
>                Session: 1098178773
>                User-Agent: openrtsp (LIVE555 Streaming Media v2009.04.20)
>
>                Thanks
>                shiv
>
>
>
>                On Sun, May 3, 2009 at 6:52 PM, Ross Finlayson
>                <finlayson at live555.com <mailto:finlayson at live555.com>
>             <mailto:finlayson at live555.com
>             <mailto:finlayson at live555.com>>> wrote:
>
>                        What do you mean "stops working"?
>                        The application stop or close automatically.. to be
>                        exactly after 5 minutes 40 sec
>
>                        I want to play this application continuously ..
>              and this
>                        the setting i am using
>
>                        openrtsp -b 500000 -v -V -w 704 -h 480  -4 -t -c
>                        rtsp://128.xxx.xx.xxx/mpeg4/media.amp >stream.m4v
>
>
>                    Please omit the "-V" option, and send us the
>             diagnostic output
>                    that "openRTSP" displays (to stderr).  This should help
>                    explain what's happening.
>
>                    --
>                    Ross Finlayson
>                    Live Networks, Inc.
>                    http://www.live555.com/
>                    _______________________________________________
>                    live-devel mailing list
>                    live-devel at lists.live555.com
>             <mailto:live-devel at lists.live555.com>
>             <mailto:live-devel at lists.live555.com
>             <mailto:live-devel at lists.live555.com>>
>
>                    http://lists.live555.com/mailman/listinfo/live-devel
>
>
>
>             ------------------------------------------------------------------------
>
>
>
>             _______________________________________________
>             live-devel mailing list
>             live-devel at lists.live555.com
>             <mailto:live-devel at lists.live555.com>
>             http://lists.live555.com/mailman/listinfo/live-devel
>              
>
>         _______________________________________________
>         live-devel mailing list
>         live-devel at lists.live555.com <mailto:live-devel at lists.live555.com>
>         http://lists.live555.com/mailman/listinfo/live-devel
>
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>   

From ganesh_vijayan at yahoo.com  Wed May  6 08:58:56 2009
From: ganesh_vijayan at yahoo.com (Ganesh V)
Date: Wed, 6 May 2009 08:58:56 -0700 (PDT)
Subject: [Live-devel] H.264 RTSP Streaming with QuickTime
Message-ID: <649529.5116.qm@web39502.mail.mud.yahoo.com>

Dear Experts,

Currently, I am trying to achieve generic RTSP H.264 Streaming functionality based on Livemedia. Based on the available literature, I figured that my application needs to generate sprop-parameter-sets for SDP. From livemedia sources, I observed that in H264VideoRTPSink (constructor) fFmtpSDPLine which contains the requisite information is generated. The pseudo-code of my application is as shown below:

    strcpy(sprop_parameter_sets,base64Encode(sps, length_of_sps);
    len = strlen(sprop_parameter_sets);
    sprop_parameter_sets[len] = ',';
    strcpy(sprop_parameter_sets+len+1, base64Encode(pps, length_of_pps));

    H264VideoRTPSink::createNew(envir(), rtpGroupsock, rtpPayloadTypeIfDynamic, p_context->profile_level_id, (const char*) sprop_parameter_sets);

With these changes, I am able to stream data from my application. This RTSP stream is successfully decoded and displayed by VLC player, which works without any issues. When I try to play this stream with Quick Time, the session is setup and the player reports "Negotiating, Buffering .." and finally, shows that QT is playing the stream, wherein the time counter is increment. However, there is no video displayed on the QT screen. The window is also not resized to the actual size of the stream and remains at the default size of the window. When I check the movie info, I observe that width, height and frame rate information is missing.

I am able to get MPEG4 streaming working without any issues with QT, but facing this issue with H.264. Has anyone tried H.264 RTSP streaming with QT prior and was successfully able to display a H.264 stream? Am I doing some mistake in my logic in invoking H264VideoRTPSink or have I missed some other initialization?

Many thanks in advance for your help.
Ganesh



      
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From wouter.dhondt at vsk.be  Wed May  6 23:56:11 2009
From: wouter.dhondt at vsk.be (Wouter Dhondt)
Date: Thu, 07 May 2009 08:56:11 +0200
Subject: [Live-devel] SET_PARAMETER & 401 due to nonce expired
Message-ID: <4A02860B.3040404@vsk.be>

Hello.

I use the livemedia client to connect to an Axis cam. All works well 
except that after a while all SET_PARAMETER requests return with 401 
Unauthorized. The stale flag is set and a new nonce value is given. At 
this point the client should re-send with the new nonce, but this 
doesn't happen. I can see that the checkForAuthenticationFailure only 
happens on DESCRIBE, OPTIONS and ANNOUNCE. How can I solve this?

Kind regards,

Wouter Dhondt



From belloni at imavis.com  Thu May  7 02:55:55 2009
From: belloni at imavis.com (Cristiano Belloni)
Date: Thu, 07 May 2009 11:55:55 +0200
Subject: [Live-devel] RTSP stops after 5 minutes
In-Reply-To: <4A014136.10202@smartt.com>
References: <4f96b010905021656v195b7086q2d4a94bdabd40506@mail.gmail.com>	<f06240801c62304d2bbd8@66.80.62.44>	<4f96b010905030835h5024da6fm8c0b2aea0b660329@mail.gmail.com>	<f06240801c623c24a0b11@66.80.62.44>	<4f96b010905031635x33ced9c2tfdb338230874911c@mail.gmail.com>	<4f96b010905031747m53fd93f7x976b46a218dde800@mail.gmail.com>	<49FE5F14.8080907@smartt.com>	<4f96b010905032204y6465ac56vfd9ff3832e8afca1@mail.gmail.com>	<4f96b010905051001u2aa3f8bbpc1c677784fe187d5@mail.gmail.com>
	<4A014136.10202@smartt.com>
Message-ID: <4A02B02B.6060406@imavis.com>

Shiva and Ron, could you please monitor the client / camera interaction 
with sniffer tools like Wireshark or Tcpdump?

The behaviour seems compatible with my very same problem (connection 
shut down after a while, only in tcp mode), and in my case it is due to 
openRTSP not sending RR packets (that causes the server to send a RTCP 
BYE after some time).

You should sniff the connection RTCP packets and see if openRTSP sends 
RR packets or, like in my case, does not at all.

Thanks,
Cristiano.

Ron McOuat wrote:
> The media.amp ending on the URL says you are using an Axis camera. If 
> you replace the -t option with -T 80 you will switch from using 
> regular RTSP/RTP by TCP to TCP tunneled on HTTP to port 80 which the 
> Axis camera will accept. This did not fail for me after a time period. 
> I compiled live555 with the -DDEBUG option added to CFLAGS and 
> followed the trace for the sessions quiting with the -t option. I 
> could not get to the bottom of it because of time pressure and I had 
> for me a workable solution using -T 80. I posted the trace to the list 
> over a year ago but as Ross says it requires getting in and tracing it 
> in detail. I also found the amount of time before failure varied up to 
> 30-40 minutes in my case. In the end I didn't know if I was fighting a 
> problem in the LAN, a problem with the Axis camera or a live555 
> problem. Axis has released patches to firmware for some models that 
> mention fixes to the MPEG4 image transfer mode so you should look at 
> that as well.
>
> Shiva Shankar N wrote:
>> Hi
>>
>>  this is there any way that i would make openrtsp client streaming to 
>> work for more than 5 minutes with TCP tunneling  i.e., with -t option.
>>
>> thanking you
>> shiv
>>
>>
>> On Mon, May 4, 2009 at 2:04 AM, Shiva Shankar N <nshamshiva at gmail.com 
>> <mailto:nshamshiva at gmail.com>> wrote:
>>
>>     Hi Ross,
>>          Thanks for your quick reply.
>>     Do you have any suggestion for me to fix this bug. I am really in
>>     urgent to finish off this project. I saw some of your previous
>>     posting in 2007 (appox april) there was a same problem. And you
>>     had release new code with the bug fixed.
>>
>>     thanks
>>     shiv
>>
>>
>>
>>     On Mon, May 4, 2009 at 12:20 AM, Ron McOuat <rmcouat at smartt.com
>>     <mailto:rmcouat at smartt.com>> wrote:
>>
>>         I ran into this with Axis cameras, judging from the URL this
>>         is what you are using. I found the -T option for HTTP
>>         tunneling also worked as well as leaving off the -t option as
>>         you have found.
>>
>>         Shiva Shankar N wrote:
>>
>>             Hi Ross,
>>
>>              It is working fine if i tunnel the data thru  UDP. I.e.,
>>             removing -t option it is working fine.
>>
>>             thanks
>>             shiv
>>
>>
>>             On Sun, May 3, 2009 at 8:35 PM, Shiva Shankar N
>>             <nshamshiva at gmail.com <mailto:nshamshiva at gmail.com>
>>             <mailto:nshamshiva at gmail.com
>>             <mailto:nshamshiva at gmail.com>>> wrote:
>>
>>                Hi Ross,
>>
>>                 This is error i am getting. please if you now where
>>             exactly it
>>                going wrong.
>>                 please let me know. So that it would reduce my
>>             debugging time
>>
>>
>>                           
>> ********************************************************************************************************************************************* 
>>
>>                User-Agent: openrtsp (LIVE555 Streaming Media 
>> v2009.04.20)
>>
>>
>>                Received OPTIONS response: RTSP/1.0 200 OK
>>                CSeq: 1
>>                Public: DESCRIBE, GET_PARAMETER, PAUSE, PLAY, SETUP,
>>             TEARDOWN
>>
>>
>>                Sending request: DESCRIBE
>>             rtsp://128.197.178.101/mpeg4/media.amp
>>             <http://128.197.178.101/mpeg4/media.amp>
>>                <http://128.197.178.101/mpeg4/media.amp> RTSP/1.0
>>
>>                CSeq: 2
>>                Accept: application/sdp
>>                User-Agent: openrtsp (LIVE555 Streaming Media 
>> v2009.04.20)
>>
>>
>>                Received DESCRIBE response: RTSP/1.0 200 OK
>>                CSeq: 2
>>                Content-Base:
>>             rtsp://128.197.178.101:554/mpeg4/media.amp/
>>             <http://128.197.178.101:554/mpeg4/media.amp/>
>>                <http://128.197.178.101:554/mpeg4/media.amp/>
>>
>>                Content-Type: application/sdp
>>                Content-Length: 684
>>
>>
>>                Need to read 684 extra bytes
>>                Read 684 extra bytes: v=0
>>                o=- 1241378493017427 1241378493017434 IN IP4
>>             128.197.178.101
>>                s=Media Presentation
>>                e=NONE
>>                c=IN IP4 0.0.0.0
>>                b=AS:8000
>>                t=0 0
>>                a=control:*
>>                a=range:npt=now-
>>                a=mpeg4-iod:
>>                           
>> "data:application/mpeg4-iod;base64,AoDUAE8BAf/1AQOAbwABQFBkYXRhOmFw
>>                           
>> cGxpY2F0aW9uL21wZWc0LW9kLWF1O2Jhc2U2NCxBUjBCR3dVZkF4Y0F5U1FBWlFRTklCRUVrK0FBZWhJ 
>>
>>                           
>> QUFIb1NBQVlCQkE9PQQNAQUABAAAAAAAAAAAAAYJAQAAAAAAAAAAAzoAAkA2ZGF0YTphcHBsaWNhdGlv 
>>
>>                           
>> bi9tcGVnNC1iaWZzLWF1O2Jhc2U2NCx3QkFTWVFTSVVFVUZQd0E9BBICDQAAAgAAAAAAAAAABQMAAEAG 
>>
>>                CQEAAAAAAAAAAA=="
>>                m=video 0 RTP/AVP 96
>>                b=AS:8000
>>                a=control:trackID=1
>>                a=rtpmap:96 MP4V-ES/90000
>>                a=fmtp:96 profile-level-id=245;
>>                config=000001B0F5000001B5090000010000000120008C4
>>                019285820F0A21F;
>>                a=mpeg4-esid:201
>>
>>                Opened URL "rtsp://128.197.178.101/mpeg4/media.amp
>>             <http://128.197.178.101/mpeg4/media.amp>
>>                <http://128.197.178.101/mpeg4/media.amp>", returning a
>>             SDP description
>>
>>                :
>>                v=0
>>                o=- 1241378493017427 1241378493017434 IN IP4
>>             128.197.178.101
>>                s=Media Presentation
>>                e=NONE
>>                c=IN IP4 0.0.0.0
>>                b=AS:8000
>>                t=0 0
>>                a=control:*
>>                a=range:npt=now-
>>                a=mpeg4-iod:
>>                           
>> "data:application/mpeg4-iod;base64,AoDUAE8BAf/1AQOAbwABQFBkYXRhOmFw
>>                           
>> cGxpY2F0aW9uL21wZWc0LW9kLWF1O2Jhc2U2NCxBUjBCR3dVZkF4Y0F5U1FBWlFRTklCRUVrK0FBZWhJ 
>>
>>                           
>> QUFIb1NBQVlCQkE9PQQNAQUABAAAAAAAAAAAAAYJAQAAAAAAAAAAAzoAAkA2ZGF0YTphcHBsaWNhdGlv 
>>
>>                           
>> bi9tcGVnNC1iaWZzLWF1O2Jhc2U2NCx3QkFTWVFTSVVFVUZQd0E9BBICDQAAAgAAAAAAAAAABQMAAEAG 
>>
>>                CQEAAAAAAAAAAA=="
>>                m=video 0 RTP/AVP 96
>>                b=AS:8000
>>                a=control:trackID=1
>>                a=rtpmap:96 MP4V-ES/90000
>>                a=fmtp:96 profile-level-id=245;
>>                config=000001B0F5000001B5090000010000000120008C4
>>                019285820F0A21F;
>>                a=mpeg4-esid:201
>>
>>                Created receiver for "video/MP4V-ES" subsession (client
>>             ports
>>                51302-51303)
>>                Sending request: SETUP
>>                rtsp://128.197.178.101:554/mpeg4/media.amp/trackID=1
>>             <http://128.197.178.101:554/mpeg4/media.amp/trackID=1>
>>                <http://128.197.178.101:554/mpeg4/media.amp/trackID=1>
>>             RTSP
>>
>>                /1.0
>>                CSeq: 3
>>                Transport: RTP/AVP/TCP;unicast;interleaved=0-1
>>                User-Agent: openrtsp (LIVE555 Streaming Media 
>> v2009.04.20)
>>
>>
>>                Received SETUP response: RTSP/1.0 200 OK
>>                CSeq: 3
>>                Session: 1098178773;timeout=60
>>                Transport:
>>             RTP/AVP/TCP;unicast;mode=play;interleaved=116-117
>>
>>
>>                Setup "video/MP4V-ES" subsession (client ports 
>> 51302-51303)
>>                Sending request: PLAY
>>             rtsp://128.197.178.101:554/mpeg4/media.amp/
>>             <http://128.197.178.101:554/mpeg4/media.amp/>
>>                <http://128.197.178.101:554/mpeg4/media.amp/> RTSP/1.0
>>
>>                CSeq: 4
>>                Session: 1098178773
>>                Range: npt=0.000-
>>                User-Agent: openrtsp (LIVE555 Streaming Media 
>> v2009.04.20)
>>
>>
>>                Received PLAY response: RTSP/1.0 200 OK
>>                CSeq: 4
>>                Session: 1098178773
>>                Range: npt=now-
>>                RTP-Info: url=trackID=1;seq=39372;rtptime=2047243131
>>
>>
>>                Started playing session
>>                Receiving streamed data...
>>                Received RTCP "BYE" on "video/MP4V-ES" subsession
>>             (after 313 seconds)
>>                Sending request: PLAY
>>             rtsp://128.197.178.101:554/mpeg4/media.amp/
>>             <http://128.197.178.101:554/mpeg4/media.amp/>
>>                <http://128.197.178.101:554/mpeg4/media.amp/> RTSP/1.0
>>
>>                CSeq: 5
>>                Session: 1098178773
>>                Range: npt=0.000-
>>                User-Agent: openrtsp (LIVE555 Streaming Media 
>> v2009.04.20)
>>
>>
>>                Failed to start playing session: Failed to read
>>             response: No error
>>                Sending request: TEARDOWN
>>                rtsp://128.197.178.101:554/mpeg4/media.amp/
>>             <http://128.197.178.101:554/mpeg4/media.amp/>
>>                <http://128.197.178.101:554/mpeg4/media.amp/> RTSP/1.0
>>
>>                CSeq: 6
>>                Session: 1098178773
>>                User-Agent: openrtsp (LIVE555 Streaming Media 
>> v2009.04.20)
>>
>>                Thanks
>>                shiv
>>
>>
>>
>>                On Sun, May 3, 2009 at 6:52 PM, Ross Finlayson
>>                <finlayson at live555.com <mailto:finlayson at live555.com>
>>             <mailto:finlayson at live555.com
>>             <mailto:finlayson at live555.com>>> wrote:
>>
>>                        What do you mean "stops working"?
>>                        The application stop or close automatically.. 
>> to be
>>                        exactly after 5 minutes 40 sec
>>
>>                        I want to play this application continuously ..
>>              and this
>>                        the setting i am using
>>
>>                        openrtsp -b 500000 -v -V -w 704 -h 480  -4 -t -c
>>                        rtsp://128.xxx.xx.xxx/mpeg4/media.amp >stream.m4v
>>
>>
>>                    Please omit the "-V" option, and send us the
>>             diagnostic output
>>                    that "openRTSP" displays (to stderr).  This should 
>> help
>>                    explain what's happening.
>>
>>                    --
>>                    Ross Finlayson
>>                    Live Networks, Inc.
>>                    http://www.live555.com/
>>                    _______________________________________________
>>                    live-devel mailing list
>>                    live-devel at lists.live555.com
>>             <mailto:live-devel at lists.live555.com>
>>             <mailto:live-devel at lists.live555.com
>>             <mailto:live-devel at lists.live555.com>>
>>
>>                    http://lists.live555.com/mailman/listinfo/live-devel
>>
>>
>>
>>             
>> ------------------------------------------------------------------------
>>
>>
>>
>>             _______________________________________________
>>             live-devel mailing list
>>             live-devel at lists.live555.com
>>             <mailto:live-devel at lists.live555.com>
>>             http://lists.live555.com/mailman/listinfo/live-devel
>>             
>>         _______________________________________________
>>         live-devel mailing list
>>         live-devel at lists.live555.com 
>> <mailto:live-devel at lists.live555.com>
>>         http://lists.live555.com/mailman/listinfo/live-devel
>>
>>
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> live-devel mailing list
>> live-devel at lists.live555.com
>> http://lists.live555.com/mailman/listinfo/live-devel
>>   
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>


-- 
Belloni Cristiano
Imavis Srl.
www.imavis.com <http://www.imavis.com>
belloni at imavis.com <mailto://belloni at imavis.com>

From alexone1980 at yahoo.it  Thu May  7 02:32:45 2009
From: alexone1980 at yahoo.it (Alex Alex)
Date: Thu, 7 May 2009 09:32:45 +0000 (GMT)
Subject: [Live-devel] Number of viewers of a multicast video stream
Message-ID: <82517.10139.qm@web23502.mail.ird.yahoo.com>

Hi,

is there a way to count how many clients are currently viewing a multicast video stream ?
If so, which classes/functions should I consider to add on a test program like testXYZVideoStreamer.cpp ?

Thanks
Alex



      
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From finlayson at live555.com  Thu May  7 03:23:59 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Thu, 7 May 2009 04:23:59 -0600
Subject: [Live-devel] SET_PARAMETER & 401 due to nonce expired
In-Reply-To: <4A02860B.3040404@vsk.be>
References: <4A02860B.3040404@vsk.be>
Message-ID: <f06240800c62866d960f0@[66.80.62.44]>

>I use the livemedia client to connect to an Axis cam. All works well 
>except that after a while all SET_PARAMETER requests return with 401 
>Unauthorized. The stale flag is set and a new nonce value is given. 
>At this point the client should re-send with the new nonce, but this 
>doesn't happen. I can see that the checkForAuthenticationFailure 
>only happens on DESCRIBE, OPTIONS and ANNOUNCE. How can I solve this?

You could add the same "checkForAuthenticationFailure" code to the 
"SET_PARAMETER" case.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From finlayson at live555.com  Thu May  7 03:37:54 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Thu, 7 May 2009 04:37:54 -0600
Subject: [Live-devel] Number of viewers of a multicast video stream
In-Reply-To: <82517.10139.qm@web23502.mail.ird.yahoo.com>
References: <82517.10139.qm@web23502.mail.ird.yahoo.com>
Message-ID: <f06240801c62869dd15e5@[66.80.62.44]>

>is there a way to count how many clients are currently viewing a 
>multicast video stream ?

Yes, we do this automatically by noting incoming RTCP "RR" packets 
from receivers.

To get this count, call "RTPSink::RTPTransmissionStatsDB::numReceivers()".
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From wouter.dhondt at vsk.be  Thu May  7 04:08:19 2009
From: wouter.dhondt at vsk.be (Wouter Dhondt)
Date: Thu, 07 May 2009 13:08:19 +0200
Subject: [Live-devel] SET_PARAMETER & 401 due to nonce expired
Message-ID: <4A02C123.70306@vsk.be>

 > You could add the same "checkForAuthenticationFailure" code to the 
"SET_PARAMETER" case.

Unfortunately I'm using the static lib so I don't build livemedia myself.



From alexone1980 at yahoo.it  Thu May  7 05:25:24 2009
From: alexone1980 at yahoo.it (Alex Alex)
Date: Thu, 7 May 2009 12:25:24 +0000 (GMT)
Subject: [Live-devel] Number of viewers of a multicast video stream
In-Reply-To: <f06240801c62869dd15e5@[66.80.62.44]>
Message-ID: <804042.64463.qm@web23505.mail.ird.yahoo.com>

thanks.
I have just tried your solution, calling videoSink->transmissionStatsDB().numReceivers() but:

1) the count is incremented only when the stream is viewed from a receiver on a different host than the streamer's one.

2) the count is not decremented when the stream is not viewed anymore from a receiver... (I just close VLC and the count remains the same)

any suggestion? 
thanks


--- Gio 7/5/09, Ross Finlayson <finlayson at live555.com> ha scritto:
Da: Ross Finlayson <finlayson at live555.com>
Oggetto: Re: [Live-devel] Number of viewers of a multicast video stream
A: "LIVE555 Streaming Media - development & use" <live-devel at ns.live555.com>
Data: Gioved? 7 maggio 2009, 10:37

> is there a way to count how many clients are currently viewing a multicast
video stream ?

Yes, we do this automatically by noting incoming RTCP "RR" packets
from receivers.

To get this count, call
"RTPSink::RTPTransmissionStatsDB::numReceivers()".
-- 
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
_______________________________________________
live-devel mailing list
live-devel at lists.live555.com
http://lists.live555.com/mailman/listinfo/live-devel



      
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From finlayson at live555.com  Thu May  7 09:52:53 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Thu, 7 May 2009 10:52:53 -0600
Subject: [Live-devel] Number of viewers of a multicast video stream
In-Reply-To: <804042.64463.qm@web23505.mail.ird.yahoo.com>
References: <804042.64463.qm@web23505.mail.ird.yahoo.com>
Message-ID: <f06240802c628c1edc655@[66.80.62.44]>

>1) the count is incremented only when the stream is viewed from a 
>receiver on a different host than the streamer's one.

Yes, but big deal...


>2) the count is not decremented when the stream is not viewed 
>anymore from a receiver...

Over time (once the record gets reclaimed) the count should get decremented.

In any case, the count can only ever be an *estimate* of the number 
of multicast receivers.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From ganesh_vijayan at yahoo.com  Thu May  7 10:17:08 2009
From: ganesh_vijayan at yahoo.com (Ganesh V)
Date: Thu, 7 May 2009 10:17:08 -0700 (PDT)
Subject: [Live-devel] H.264 RTSP Streaming with QuickTime
In-Reply-To: <649529.5116.qm@web39502.mail.mud.yahoo.com>
References: <649529.5116.qm@web39502.mail.mud.yahoo.com>
Message-ID: <845860.60257.qm@web39505.mail.mud.yahoo.com>

I got H.264 Streaming working with Quicktime. While preparing the sprop_parameter_sets as shown below, I shouldn't have been including the NALU start code which was the problem.

Currently, quicktime is able to identify the incoming stream and is able to set it's window width and height. However, I observe a green screen and unable to view the video. Has anyone faced this issue before and solved the same? 

Thanks.




________________________________
From: Ganesh V <ganesh_vijayan at yahoo.com>
To: live-devel at ns.live555.com
Sent: Wednesday, May 6, 2009 9:28:56 PM
Subject: [Live-devel] H.264 RTSP Streaming with QuickTime


Dear Experts,

Currently, I am trying to achieve generic RTSP H.264 Streaming functionality based on Livemedia. Based on the available literature, I figured that my application needs to generate sprop-parameter-sets for SDP. From livemedia sources, I observed that in H264VideoRTPSink (constructor) fFmtpSDPLine which contains the requisite information is generated. The pseudo-code of my application is as shown below:

    strcpy(sprop_parameter_sets,base64Encode(sps, length_of_sps);
    len = strlen(sprop_parameter_sets);
    sprop_parameter_sets[len] = ',';
    strcpy(sprop_parameter_sets+len+1, base64Encode(pps, length_of_pps));

    H264VideoRTPSink::createNew(envir(), rtpGroupsock, rtpPayloadTypeIfDynamic, p_context->profile_level_id, (const char*) sprop_parameter_sets);

With these changes, I am able to stream data from my application. This RTSP stream is successfully decoded and displayed by VLC player, which works without any issues. When I try to play this stream with Quick Time, the session is setup and the player reports "Negotiating, Buffering .." and finally, shows that QT is playing the stream, wherein the time counter is increment. However, there is no video displayed on the QT screen. The window is also not resized to the actual size of the stream and remains at the default size of the window. When I check the movie info, I observe that width, height and frame rate information is missing.

I am able to get MPEG4 streaming working without any issues with QT, but facing this issue with H.264. Has anyone tried H.264 RTSP streaming with QT prior and was successfully able to display a H.264 stream? Am I doing some mistake in my logic in invoking H264VideoRTPSink or have I missed some other initialization?

Many thanks in advance for your help.
Ganesh


      
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From kidjan at gmail.com  Fri May  8 08:50:41 2009
From: kidjan at gmail.com (Jeremy Noring)
Date: Fri, 8 May 2009 08:50:41 -0700
Subject: [Live-devel] H.264 RTSP Streaming with QuickTime
In-Reply-To: <845860.60257.qm@web39505.mail.mud.yahoo.com>
References: <649529.5116.qm@web39502.mail.mud.yahoo.com>
	<845860.60257.qm@web39505.mail.mud.yahoo.com>
Message-ID: <bda4ea420905080850y19954640o634a58abfc04e8c3@mail.gmail.com>

I've gotten about as far.  I don't get any green video, but QT does
determine my video width and height and start streaming.  I haven't managed
to figure out anything beyond that, but admittedly I haven't looked into it
for a few weeks.

On Thu, May 7, 2009 at 10:17 AM, Ganesh V <ganesh_vijayan at yahoo.com> wrote:

> I got H.264 Streaming working with Quicktime. While preparing the
> sprop_parameter_sets as shown below, I shouldn't have been including the
> NALU start code which was the problem.
>
> Currently, quicktime is able to identify the incoming stream and is able to
> set it's window width and height. However, I observe a green screen and
> unable to view the video. Has anyone faced this issue before and solved the
> same?
>
> Thanks.
>
> ------------------------------
> *From:* Ganesh V <ganesh_vijayan at yahoo.com>
> *To:* live-devel at ns.live555.com
> *Sent:* Wednesday, May 6, 2009 9:28:56 PM
> *Subject:* [Live-devel] H.264 RTSP Streaming with QuickTime
>
> Dear Experts,
>
> Currently, I am trying to achieve generic RTSP H.264 Streaming
> functionality based on Livemedia. Based on the available literature, I
> figured that my application needs to generate sprop-parameter-sets for SDP.
> From livemedia sources, I observed that in H264VideoRTPSink (constructor)
> fFmtpSDPLine which contains the requisite information is generated. The
> pseudo-code of my application is as shown below:
>
>     strcpy(sprop_parameter_sets,base64Encode(sps, length_of_sps);
>     len = strlen(sprop_parameter_sets);
>     sprop_parameter_sets[len] = ',';
>     strcpy(sprop_parameter_sets+len+1, base64Encode(pps, length_of_pps));
>
>     H264VideoRTPSink::createNew(envir(), rtpGroupsock,
> rtpPayloadTypeIfDynamic, p_context->profile_level_id, (const char*)
> sprop_parameter_sets);
>
> With these changes, I am able to stream data from my application. This RTSP
> stream is successfully decoded and displayed by VLC player, which works
> without any issues. When I try to play this stream with Quick Time, the
> session is setup and the player reports "Negotiating, Buffering .." and
> finally, shows that QT is playing the stream, wherein the time counter is
> increment. However, there is no video displayed on the QT screen. The window
> is also not resized to the actual size of the stream and remains at the
> default size of the window. When I check the movie info, I observe that
> width, height and frame rate information is missing.
>
> I am able to get MPEG4 streaming working without any issues with QT, but
> facing this issue with H.264. Has anyone tried H.264 RTSP streaming with QT
> prior and was successfully able to display a H.264 stream? Am I doing some
> mistake in my logic in invoking H264VideoRTPSink or have I missed some other
> initialization?
>
> Many thanks in advance for your help.
> Ganesh
>
>
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
>


-- 
Where are we going?
And why am I in this hand-basket?
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From michael at abriskwok.net  Fri May  8 13:36:12 2009
From: michael at abriskwok.net (Michael Barkowski)
Date: Fri, 8 May 2009 16:36:12 -0400
Subject: [Live-devel] MPEG-4 Streaming presentation rate
Message-ID: <e37942f90905081336i534aef0au771df67e686597b8@mail.gmail.com>

Hello all,

I'm new to this forum, so please take a moment to let me know if I'm
not posing this question in the right way.

I compiled the Livemedia April 20, 2009 release.

I'm trying to use testOnDemandRTSPServer test program to stream MPEG-4
to an Mplayer client. The two machines are connected via a very fast
Gigabit Ethernet link.

It seems to be playing at about half the desired frame rate.  The
desired frame rate is 30 fps.  Mplayer is invoked with -fps 30 to no
effect.

If I stream to Mencoder, and play the saved file using Mplayer, it
plays at the right speed.

The file I am using as a source to testOnDemandRTSP Server was
captured by either openRTSP (using -4 to output to Quicktime
container) or Mencoder using straight copy (two cases with the same
symptom) from an Axis M1101 camera.

CPU utilization is low for all activities mentioned.

I have tried changing the default frame rate in playCommon.cpp which
is used by openRTSP.
-unsigned movieFPS = 15; // default
+unsigned movieFPS = 30; // default

This has no effect.

Any ideas?  What is testOnDemandRTSP Server missing?  Just the frame rate?

--
Michael Barkowski

From swolfod at hotmail.com  Sun May 10 09:28:13 2009
From: swolfod at hotmail.com (Lindsey Liao)
Date: Mon, 11 May 2009 00:28:13 +0800
Subject: [Live-devel] RTSP streaming
Message-ID: <BAY113-W339D5A55EBE0A072ECCD2A5620@phx.gbl>


Hi everyone,

 

I am totally new of live555 or RTSP. Currently I am studying how to use live555 to build a RTSP server. I tried out the 'testOnDemandRTSPServer.exe' application, and it successfully setup an RTSP server which can be visited by other machine in the local LAN. However, when I tried to visit the server from the WAN, that is, from machines outside the router, it appears that the server is not receiving any request.

 

Can someone tell me what is the reason to this problem, and is there any workaround?

 

Thanks,
Lindsey Liao



_________________________________________________________________
Show them the way! Add maps and directions to your party invites. 
http://www.microsoft.com/windows/windowslive/products/events.aspx
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From dcharmet at uniways.fr  Mon May 11 00:33:06 2009
From: dcharmet at uniways.fr (Denis Charmet)
Date: Mon, 11 May 2009 09:33:06 +0200
Subject: [Live-devel] Infinite loop in liveMedia/MediaSession.cpp with
	Windows CE
Message-ID: <4A07D4B2.20209@uniways.fr>

Hello,

I'd like to report a bug in MediaSubsession::initiate. Windows CE/Mobile 
shitty IPstack sometimes seems to always return the same odd port so 
initiate() loops and creates sockets until the device hasn't enough memory.

The Hashtable mechanism won't work as the result socketHashTable->Add() 
isn't checked. If the port was already used it returns the old value but 
it won't be freed.

Lastely, the Hashtable is only released when the operation successes.

I hope this can help and I'm working on a patch quite similar to Pierre 
Ynard's one.

Best regards.

-- 
Denis Charmet

From wouter.dhondt at vsk.be  Mon May 11 01:11:31 2009
From: wouter.dhondt at vsk.be (Wouter Dhondt)
Date: Mon, 11 May 2009 10:11:31 +0200
Subject: [Live-devel] Shared lib instead of static
Message-ID: <4A07DDB3.70900@vsk.be>

Is there an easy way to change the config.linux so that the result is a 
shared lib (like the one from liblivemedia-dev) ?



From finlayson at live555.com  Mon May 11 01:22:17 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Mon, 11 May 2009 02:22:17 -0600
Subject: [Live-devel] RTSP streaming
In-Reply-To: <BAY113-W339D5A55EBE0A072ECCD2A5620@phx.gbl>
References: <BAY113-W339D5A55EBE0A072ECCD2A5620@phx.gbl>
Message-ID: <f06240800c62d907f9eaa@[66.80.62.44]>

>Hi everyone,
>
>I am totally new of live555 or RTSP. Currently I am studying how to 
>use live555 to build a RTSP server. I tried out the 
>'testOnDemandRTSPServer.exe' application, and it successfully setup 
>an RTSP server which can be visited by other machine in the local 
>LAN. However, when I tried to visit the server from the WAN, that 
>is, from machines outside the router, it appears that the server is 
>not receiving any request.
>
>Can someone tell me what is the reason to this problem

You have a firewall somewhere (between your client and server) that 
is blocking RTP/UDP packets.

>, and is there any workaround?

Fix your firewall.  Failing that, try having your client use RTP-over-TCP.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From finlayson at live555.com  Mon May 11 01:31:29 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Mon, 11 May 2009 02:31:29 -0600
Subject: [Live-devel] Infinite loop in liveMedia/MediaSession.cpp with
 Windows CE
In-Reply-To: <4A07D4B2.20209@uniways.fr>
References: <4A07D4B2.20209@uniways.fr>
Message-ID: <f06240802c62d91fef8b4@[66.80.62.44]>

>I'd like to report a bug in MediaSubsession::initiate. Windows 
>CE/Mobile shitty IPstack sometimes seems to always return the same 
>odd port so initiate() loops and creates sockets until the device 
>hasn't enough memory.
>
>The Hashtable mechanism won't work as the result 
>socketHashTable->Add() isn't checked. If the port was already used 
>it returns the old value but it won't be freed.

But if the port was allocated the first time (and recorded in the 
hash table), then your OS should never be allocating it (as an 
ephemeral port) a second time!  If your OS really is doing this 
(allocating the same ephemeral port number a second time, while the 
first-allocated port is still in use), then it is badly broken, and 
our code has no chance of working with it.  If this is really 
happening, you will need to fix (or replace) your OS.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From finlayson at live555.com  Mon May 11 02:07:19 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Mon, 11 May 2009 03:07:19 -0600
Subject: [Live-devel] MPEG-4 Streaming presentation rate
In-Reply-To: <e37942f90905081336i534aef0au771df67e686597b8@mail.gmail.com>
References: <e37942f90905081336i534aef0au771df67e686597b8@mail.gmail.com>
Message-ID: <f06240804c62d9acd093c@[66.80.62.44]>

>The file I am using as a source to testOnDemandRTSP Server was
>captured by either openRTSP (using -4 to output to Quicktime
>container)

Our server *cannot* stream from this type of file (".mp4" or ".mov" 
format).  The only type of MPEG-4 file it can stream from is a MPEG-4 
Video Elementary Stream.  (If you want to record files of this type 
using "openRTSP", then *do not* use the "-4" or "-q" options.)
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From linkfanel at yahoo.fr  Mon May 11 03:37:21 2009
From: linkfanel at yahoo.fr (Pierre Ynard)
Date: Mon, 11 May 2009 12:37:21 +0200
Subject: [Live-devel] Infinite loop in liveMedia/MediaSession.cpp with
	Windows CE
Message-ID: <20090511103721.GA4419@via.ecp.fr>

Hello,

> But if the port was allocated the first time (and recorded in the 
> hash table), then your OS should never be allocating it (as an 
> ephemeral port) a second time!  If your OS really is doing this 
> (allocating the same ephemeral port number a second time, while the 
> first-allocated port is still in use), then it is badly broken, and 
> our code has no chance of working with it.  If this is really 
> happening, you will need to fix (or replace) your OS.

I work with Denis, and this is the same problem that happened in last
December
(http://lists.live555.com/pipermail/live-devel/2008-December/009967.html).
Yes indeed, this time, just as before, the same source port is returned
by WinCE. At that time, manually incrementing the port number did the
trick. Now that a more sophiscated hash table has implemented, manual
handling of the port number is gone, and so is the work-around for
WinCE, thus the recent breakage.

I agree that WinCE should not be doing that. However, IMHO, requesting
a port over and over, assuming that the OS will eventually give you
something suitable, is a risky behavior. The socket API doesn't
guarantee that it will return an even port in a quick and efficient way
for your application, it might as well try and exhaust all 30000+ odd
ports first, making the code slow, clogging the hash table, leading to
memory failure, EMFILE errors...

So I think that it would be appropriate for live555 code to implement
some measures making sure that this doesn't happen, and while we're at
it, possibly working around OS bugs.

Kind regards,

-- 
Pierre Ynard
"Une ?me dans un corps, c'est comme un dessin sur une feuille de papier."

From finlayson at live555.com  Mon May 11 06:15:36 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Mon, 11 May 2009 07:15:36 -0600
Subject: [Live-devel] Infinite loop in liveMedia/MediaSession.cpp with
 Windows CE
In-Reply-To: <20090511103721.GA4419@via.ecp.fr>
References: <20090511103721.GA4419@via.ecp.fr>
Message-ID: <f06240806c62dd289c81d@[66.80.62.44]>

>I agree that WinCE should not be doing that. However, IMHO, requesting
>a port over and over, assuming that the OS will eventually give you
>something suitable, is a risky behavior

There's really no alternative.  We can't just arbitrarily choose an 
even number ourselves, in case we get a port number that happens to 
be used by someone else.  And we also have to make sure that - if the 
code is run in more than one process (i.e., application) on the same 
host - we don't end up with more than one process accidentally using 
the same port number.


>So I think that it would be appropriate for live555 code to implement
>some measures making sure that this doesn't happen, and while we're at
>it, possibly working around OS bugs.

No, in general we're not going to "work around OS bugs" - especially 
ones as aggregious as this.  What I will do, however, is change the 
code - where it "Add()"s a record for a port number to the hash table 
- to check whether there was already a hash table entry (for the same 
port number) there - so at least we don't end up with a memory leak 
in this situation.

However, I find it difficult to believe that any OS - even WinCE - is 
really allocating the same emphemeral port number more than once in a 
row (while the first allocated port is still in use).  If this is 
really happening, then there must surely be a bugfix OS upgrade 
available, so you need to apply that first.  Sorry.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From dcharmet at uniways.fr  Mon May 11 06:52:43 2009
From: dcharmet at uniways.fr (Denis Charmet)
Date: Mon, 11 May 2009 15:52:43 +0200
Subject: [Live-devel] Infinite loop in liveMedia/MediaSession.cpp with
 Windows CE
In-Reply-To: <f06240806c62dd289c81d@[66.80.62.44]>
References: <20090511103721.GA4419@via.ecp.fr>
	<f06240806c62dd289c81d@[66.80.62.44]>
Message-ID: <4A082DAB.7070808@uniways.fr>

Ross Finlayson wrote:
>> I agree that WinCE should not be doing that. However, IMHO, requesting
>> a port over and over, assuming that the OS will eventually give you
>> something suitable, is a risky behavior
>
> There's really no alternative.  We can't just arbitrarily choose an 
> even number ourselves, in case we get a port number that happens to be 
> used by someone else.  And we also have to make sure that - if the 
> code is run in more than one process (i.e., application) on the same 
> host - we don't end up with more than one process accidentally using 
> the same port number.
>
>
>> So I think that it would be appropriate for live555 code to implement
>> some measures making sure that this doesn't happen, and while we're at
>> it, possibly working around OS bugs.
>
> No, in general we're not going to "work around OS bugs" - especially 
> ones as aggregious as this.  What I will do, however, is change the 
> code - where it "Add()"s a record for a port number to the hash table 
> - to check whether there was already a hash table entry (for the same 
> port number) there - so at least we don't end up with a memory leak in 
> this situation.
There is also the 'failure' memory leak:

if (!success) break;  // a fatal error occurred trying to create the RTP 
and RTCP sockets; we can't continue    (liveMedia/MediaSession.cpp : 685)

which IMHO could become:

if (!success) {
  Groupsock* oldGS;
  while ((oldGS = (Groupsock*)socketHashTable->RemoveNext()) != NULL) {
    delete oldGS;
  }
  delete socketHashTable;
  break; // a fatal error occurred trying to create the RTP and RTCP 
sockets; we can't continue
}


By the way Microsoft's excuse for that is that getsockname can return 
false informations before the socket is connected as devices can be 
multihomed.
http://msdn.microsoft.com/en-us/library/ms738543(VS.85).aspx


Thanks for the answers.

Best regards.

-- 
Denis Charmet

From finlayson at live555.com  Mon May 11 07:57:58 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Mon, 11 May 2009 08:57:58 -0600
Subject: [Live-devel] Infinite loop in liveMedia/MediaSession.cpp with
 Windows CE
In-Reply-To: <4A082DAB.7070808@uniways.fr>
References: <20090511103721.GA4419@via.ecp.fr>
	<f06240806c62dd289c81d@[66.80.62.44]> <4A082DAB.7070808@uniways.fr>
Message-ID: <f06240807c62ded5476a3@[66.80.62.44]>

>There is also the 'failure' memory leak

Yes, I'll take care of that as well.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From mrteddy at citromail.hu  Mon May 11 17:34:38 2009
From: mrteddy at citromail.hu (Mr. Teddy)
Date: Tue, 12 May 2009 02:34:38 +0200
Subject: [Live-devel] openRTSP TS receive as an another program input
In-Reply-To: <f06240801c625efc348ae@[66.80.62.44]>
Message-ID: <20090512003438.16060.qmail@server16.citromail.hu>

            Hello!
 
When I use this settings openRTSP immeditaly sends TEARDOWN request, and the receiving stops, so my application can't read from stdin, because the openRTSP stop.
 
Best Regards:
Peter
-- Eredeti ?zenet --
Felad?: Ross Finlayson &lt;finlayson at live555.com&gt;
C?mzett: LIVE555 Streaming Media - development &amp; use &lt;live-devel at ns.live555.com&gt;
M?solat: 
Elk?ldve: 2009.05.05  15:51
T?ma: Re: [Live-devel] openRTSP TS receive as an another program input&gt;To pipe openRTSP don't you have a little sample code? I tried &gt;_popen() function, but it not the best.No, you don't need to write any code for this; you can do this from the command lineopenRTSP -v etc. rtsp://url | your-application-that-reads-from-stdin-- Ross FinlaysonLive Networks, Inc.http://www.live555.com/_______________________________________________live-devel mailing listlive-devel at lists.live555.comhttp://lists.live555.com/mailman/listinfo/live-devel


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From andy at j2kvideo.com  Tue May 12 08:43:39 2009
From: andy at j2kvideo.com (Andy Bell)
Date: Tue, 12 May 2009 17:43:39 +0200
Subject: [Live-devel] Presentation time in the future.
Message-ID: <a7a356890905120843n25e33aadt60bcfea877160147@mail.gmail.com>

Hi,

I have implemented a media sink for an RTSP video stream using Live555.  All
works great expect that the presentation time is 2 hours ahead of local
time, why is this?  If we call gettimeofday() on the same machine we get the
correct time.


Thanks in advance.
-- 
Andy Bell
CTO
J2K Video Limited

T: +44 (0)20 8133 2473
M: +34 685 130097
E: andy at j2kvideo.com
W: www.j2kvideo.com
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From michael at abriskwok.net  Tue May 12 08:31:08 2009
From: michael at abriskwok.net (Michael Barkowski)
Date: Tue, 12 May 2009 11:31:08 -0400
Subject: [Live-devel] MPEG-4 Streaming presentation rate
In-Reply-To: <f06240804c62d9acd093c@66.80.62.44>
References: <e37942f90905081336i534aef0au771df67e686597b8@mail.gmail.com>
	<f06240804c62d9acd093c@66.80.62.44>
Message-ID: <e37942f90905120831y1110d31au45fb17a95fc977d4@mail.gmail.com>

Hello Ross,

On Mon, May 11, 2009 at 5:07 AM, Ross Finlayson <finlayson at live555.com> wrote:
>> The file I am using as a source to testOnDemandRTSP Server was
>> captured by either openRTSP (using -4 to output to Quicktime
>> container)
>
> Our server *cannot* stream from this type of file (".mp4" or ".mov" format).
>  The only type of MPEG-4 file it can stream from is a MPEG-4 Video
> Elementary Stream.  (If you want to record files of this type using
> "openRTSP", then *do not* use the "-4" or "-q" options.)

Thanks so much for the reply.

Ok - I've now tried with openRTSP creating MPEG-4 ES - no "-4" or "-q".
Having the same issue.

Turned on debug...  testOnDemandRTSP is calculating presentation times
that work out to 15 fps, for example:
#parsing VideoObjectPlane
vop_coding_type: 1(P), modulo_time_base: 0, vop_time_increment: 133
MPEGVideoStreamFramer::computePresentationTime(133) -> 98426.157286 [0.240799]
fDurationInMicroseconds: 67000 ((67*1000000)/1000.000000)

The file should be 30 fps, and plays as such in VLC.

If I change the camera settings to 15 fps, then everything is 15 fps
and testOnDemandRTSP streams things at the proper rate.

How do I go about checking the file that openRTSP has created?  Where
do I look to find out why testOnDemandRTSP always calculates 15 fps?

Many thanks,

--
Michael Barkowski

From bourdag at gmail.com  Tue May 12 08:49:11 2009
From: bourdag at gmail.com (bourda guillaume)
Date: Tue, 12 May 2009 17:49:11 +0200
Subject: [Live-devel] openRTSP - audio file without header ?
Message-ID: <c0a55910905120849hcd74764jd16238111508fd2e@mail.gmail.com>

Hi everybody,

I am using openRTSP to record the video and the audio stream of an RTSP
source (openRTSP rtsp://url).
As expected, openRTSP is creating two files: "audio-MPEG4-GENERIC-2" for the
audio (aac) and "video-MP4V-ES-1" for the video (m4v).
I manage to play the video file with vlc (vlc file/m4v://video-MP4V-ES-1),
however vlc fails when I try to do the same on the audio file (vlc
file/m4a://audio-MPEG4-GENERIC-2).

Here bellow is the output of vlc in verbose mode:
############
[00000305] main packetizer debug: using packetizer module
"packetizer_mpeg4audio"
[00000303] main demuxer debug: using demux2 module "m4a"
[00000298] main input debug: thread 1126287696 (input) created at priority 0
(input/input.c:265)
[00000298] main input debug: `file/m4a://audio-MPEG4-GENERIC-2' successfully
opened
[00000298] main input debug: EOF reached
[00000298] main input debug: closing input
############

I believe this is because openRTSP creates a "raw" aac file, i.e a file
without header.

Thus my questions:
How can I change openRTSP so that it creates a header for the audio file?
Is it possible to use the SDP information to create a header?
Does anyone knows a program/library that I can use to create a header for my
aac raw file(ex: from SDP info)?


I am a newbe to openRTSP, thanks a lot in advance for any help!
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From finlayson at live555.com  Tue May 12 09:21:21 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Tue, 12 May 2009 09:21:21 -0700
Subject: [Live-devel] MPEG-4 Streaming presentation rate
In-Reply-To: <e37942f90905120831y1110d31au45fb17a95fc977d4@mail.gmail.com>
References: <e37942f90905081336i534aef0au771df67e686597b8@mail.gmail.com>
	<f06240804c62d9acd093c@66.80.62.44>
	<e37942f90905120831y1110d31au45fb17a95fc977d4@mail.gmail.com>
Message-ID: <f06240805c62f51a2c90a@[66.80.62.44]>

>Ok - I've now tried with openRTSP creating MPEG-4 ES - no "-4" or "-q".
>Having the same issue.
>
>Turned on debug...  testOnDemandRTSP is calculating presentation times
>that work out to 15 fps, for example:
>#parsing VideoObjectPlane
>vop_coding_type: 1(P), modulo_time_base: 0, vop_time_increment: 133
>MPEGVideoStreamFramer::computePresentationTime(133) -> 98426.157286 [0.240799]
>fDurationInMicroseconds: 67000 ((67*1000000)/1000.000000)
>
>The file should be 30 fps, and plays as such in VLC.

The frame rate of the outgoing stream is set (in this case) at line 
413 of "MPEG4VideoStreamFramer.cpp".  You'll need to examine this 
code to figure out why it's not working properly for your data.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From finlayson at live555.com  Tue May 12 09:24:30 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Tue, 12 May 2009 09:24:30 -0700
Subject: [Live-devel] openRTSP - audio file without header ?
In-Reply-To: <c0a55910905120849hcd74764jd16238111508fd2e@mail.gmail.com>
References: <c0a55910905120849hcd74764jd16238111508fd2e@mail.gmail.com>
Message-ID: <f06240806c62f52bf0c04@[66.80.62.44]>

>I believe this is because openRTSP creates a "raw" aac file, i.e a 
>file without header.
>
>Thus my questions:
>How can I change openRTSP so that it creates a header for the audio file?

You would need to write a new 'file sink' class for AAC audio files 
with headers, and use this in "openRTSP" (in 
"testProgs/playCommon.cpp").

For illustration, note how we do something similar when writing H.264 
video files.  Note the "H264VideoFileSink" class, and it's use in the 
"openRTSP" code.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From finlayson at live555.com  Tue May 12 09:27:38 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Tue, 12 May 2009 09:27:38 -0700
Subject: [Live-devel] Presentation time in the future.
In-Reply-To: <a7a356890905120843n25e33aadt60bcfea877160147@mail.gmail.com>
References: <a7a356890905120843n25e33aadt60bcfea877160147@mail.gmail.com>
Message-ID: <f06240807c62f534d2d0e@[66.80.62.44]>

>I have implemented a media sink for an RTSP video stream using 
>Live555.  All works great expect that the presentation time is 2 
>hours ahead of local time, why is this?

You tell me.  You are (or at least, should be) the one who sets the 
presentation time in the 'data source' class that feeds into your 
'sink' object.  You should be setting "fPresentationTime" in your 
implementation of "doGetNextFrame()" in your 'data source' class. 
(See, for example, the comments in "DeviceSource.cpp".)

-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From ganesh_vijayan at yahoo.com  Sat May 16 05:07:17 2009
From: ganesh_vijayan at yahoo.com (Ganesh V)
Date: Sat, 16 May 2009 05:07:17 -0700 (PDT)
Subject: [Live-devel] H.264 RTSP Streaming with QuickTime
Message-ID: <493407.53809.qm@web39503.mail.mud.yahoo.com>

I tried one more experiment couple of days back. I dumped a stream prepared for RTSP streaming using QT Pro and analyzed the same. When I analyzed the stream, I found that after sprop_parameter_sets, some elements of "hinf" atom from QTFF were present in the stream. I am not sure if this is really required for achieving H.264 display on the screen, but this was something different.

Has anyone faced/observed similar issues previously?



      
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From finlayson at live555.com  Sat May 16 06:57:45 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Sat, 16 May 2009 06:57:45 -0700
Subject: [Live-devel] H.264 RTSP Streaming with QuickTime
In-Reply-To: <493407.53809.qm@web39503.mail.mud.yahoo.com>
References: <493407.53809.qm@web39503.mail.mud.yahoo.com>
Message-ID: <f06240804c634766a1c0f@[66.80.62.44]>

>I tried one more experiment couple of days back. I dumped a stream 
>prepared for RTSP streaming using QT Pro and analyzed the same. When 
>I analyzed the stream, I found that after sprop_parameter_sets, some 
>elements of "hinf" atom from QTFF were present in the stream. I am 
>not sure if this is really required for achieving H.264 display on 
>the screen

No, they're not.  The "hinf" atoms are for 'hinting', which is a 
trick used only by Apple's "QuickTime Streaming Server" when 
streaming the file.  They have nothing to do with playing the file, 
and are not required for playing the file.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From ottavio at videotec.com  Mon May 18 02:53:43 2009
From: ottavio at videotec.com (Ottavio Campana)
Date: Mon, 18 May 2009 11:53:43 +0200
Subject: [Live-devel] doubt about H264VideoRTPSink
Message-ID: <4A113027.1050405@videotec.com>

Hi,

I'm trying to stream an H264 file with livemedia, but I'm having a
problem with the framerate.

By using wireshark, I've seen that the stream is declare to have a clock
period of 0.9 ms, thus giving approx 11 fps.

The response to the describe command is

RTSP/1.0 200 OK
CSeq: 2
Date: Mon, May 18 2009 07:20:24 GMT
Content-Base: rtsp://192.168.0.1:8554/testStream/
Content-Type: application/sdp
Content-Length: 605

v=0
o=- 1242631222767144 1 IN IP4 192.168.0.1
s=Session streamed by "testMPEG4VideoStreamer"
i=/home/ottavio/Projects/JM/bin/test.264
t=0 0
a=tool:LIVE555 Streaming Media v2007.02.20
a=type:broadcast
a=control:*
a=source-filter: incl IN IP4 * 192.168.0.1
a=rtcp-unicast: reflection
a=range:npt=0-
a=x-qt-text-nam:Session streamed by "testMPEG4VideoStreamer"
a=x-qt-text-inf:/home/ottavio/Projects/JM/bin/test.264
m=video 18888 RTP/AVP 96
c=IN IP4 232.180.62.232/255
a=rtpmap:96 H264/90000
a=fmtp:96
packetization-mode=1;profile-level-id=000042;sprop-parameter-sets=h264
a=control:track1

so I think I have to change the line "a=rtpmap:96 H264/90000" into
something like "a=rtpmap:96 H264/40000" if I want to achieve 25 fps. But
the question is, how do I change it?

If I give a look at the source of H264VideoRTPSink, I find:

H264VideoRTPSink
::H264VideoRTPSink(UsageEnvironment& env, Groupsock* RTPgs,
                   unsigned char rtpPayloadFormat,
                   unsigned profile_level_id,
                   char const* sprop_parameter_sets_str)
  : VideoRTPSink(env, RTPgs, rtpPayloadFormat, 90000, "H264"),
    fOurFragmenter(NULL) {

so 90000 seems to be hard coded.

Am I missing something? Or how do I change it?

Thanks for your help,

Ottavio

From ldacvs at gmail.com  Mon May 18 05:53:27 2009
From: ldacvs at gmail.com (ldac)
Date: Mon, 18 May 2009 14:53:27 +0200
Subject: [Live-devel] H264 live streaming (multicast)
Message-ID: <4A115A47.4060803@gmail.com>


    Hi everybody,

I am new using Live555 and I have the following question: What should I 
do to stream live video (H264, from a web cam) to a STB Amino 130 (using 
multicast)? Is there some Live555 test program to do that?

    Thank you very much in advance, all the best.
    LD.

From finlayson at live555.com  Mon May 18 07:51:07 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Mon, 18 May 2009 07:51:07 -0700
Subject: [Live-devel] doubt about H264VideoRTPSink
In-Reply-To: <4A113027.1050405@videotec.com>
References: <4A113027.1050405@videotec.com>
Message-ID: <f06240800c63720b8a6c3@[66.80.62.44]>

>I'm trying to stream an H264 file with livemedia, but I'm having a
>problem with the framerate.

The rate at which outgoing stream data gets sent is determined by the 
object (or chain of objects) that feeds into the "H264VideoRTPSink", 
not the "H264VideoRTPSink" itself.  In particular, make sure that 
you are setting the "fDurationInMicroseconds" and "fPresentationTime" 
parameters correctly.


>so I think I have to change the line "a=rtpmap:96 H264/90000"

No - do not change this!  The 90000 is the RTP timestamp frequency. 
For MPEG video streams (including H.264), this is always 90000.  It 
has nothing to do with the video frame rate, or the rate at which 
data gets transmitted.


>so 90000 seems to be hard coded.

Yes, see above.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From ottavio at videotec.com  Mon May 18 09:37:00 2009
From: ottavio at videotec.com (Ottavio Campana)
Date: Mon, 18 May 2009 18:37:00 +0200
Subject: [Live-devel] about streaming plain h264
Message-ID: <4A118EAC.6000602@videotec.com>

I'm trying to stream a video-only annex-b H264 file with RTP. I don't 
understand why, but it only seams to be working with mplyer and not vlc.

I know there's not ready to use solution for h.264/avc in libvemedia, 
but do you have a working example streaming NAL units that I can use as 
example?

By reading the list archives, I found

http://www.mail-archive.com/live-devel at lists.live555.com/msg03410.html
http://www.mail-archive.com/live-devel at lists.live555.com/msg03412.html

But it does not seam to work with live 2009.04.20 . I hope Robert 
Klotzner can give me a hint....

Thanks.

From finlayson at live555.com  Mon May 18 15:57:59 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Mon, 18 May 2009 15:57:59 -0700
Subject: [Live-devel] about streaming plain h264
In-Reply-To: <4A118EAC.6000602@videotec.com>
References: <4A118EAC.6000602@videotec.com>
Message-ID: <f06240800c637982663c9@[66.80.62.44]>

>I'm trying to stream a video-only annex-b H264 file with RTP. I 
>don't understand why, but it only seams to be working with mplyer 
>and not vlc.

Have you asked about this on the VLC mailing list (vlc at videolan.org)? 
Both MPlayer and VLC use our RTSP/RTP client software, so whatever's 
causing VLC problems might have nothing to do with us.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From bitter at vtilt.com  Mon May 18 16:50:04 2009
From: bitter at vtilt.com (Brad Bitterman)
Date: Mon, 18 May 2009 19:50:04 -0400
Subject: [Live-devel] non-blocking describe in RTSPClient?
Message-ID: <55791956-5835-427A-86CA-6B9411F14D43@vtilt.com>

Is there a non-blocking Describe implementation for RTSPClient? I have  
multiple clients and if I bring one online that has a server that is  
not available, RTSPClient blocks the other running clients until the  
timeout occurs.

I saw a partial post that was made in 2007 regarding the issue but  
could not make much sense of it...

Thanks,
Brad

From finlayson at live555.com  Mon May 18 17:04:51 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Mon, 18 May 2009 17:04:51 -0700
Subject: [Live-devel] non-blocking describe in RTSPClient?
In-Reply-To: <55791956-5835-427A-86CA-6B9411F14D43@vtilt.com>
References: <55791956-5835-427A-86CA-6B9411F14D43@vtilt.com>
Message-ID: <f06240802c637a788fe79@[66.80.62.44]>

>Is there a non-blocking Describe implementation for RTSPClient?

Not yet, but for now (pending a complete reimplementation of 
"RTSPClient", which is coming) you can use the "timeout" parameter to 
"describeURL()".  This allows you to specify an upper bound (in 
seconds) on how long to block.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From renatomauro at libero.it  Mon May 18 18:03:25 2009
From: renatomauro at libero.it (Renato MAURO (Libero))
Date: Tue, 19 May 2009 03:03:25 +0200
Subject: [Live-devel] NetCommon.hh and UsageEnvironment.hh
References: <55791956-5835-427A-86CA-6B9411F14D43@vtilt.com>
	<f06240802c637a788fe79@[66.80.62.44]>
Message-ID: <5E7AC58DE2EC40EAAFC5263745941DB4@CSystemDev>

Hello Ross.

    Could you confirm that NetCommon.hh is included in every cpp file of 
Live555? And UsageEnvironment.hh?

Thank you,

    Renato MAURO


----- Original Message ----- 
From: "Ross Finlayson" <finlayson at live555.com>
To: "LIVE555 Streaming Media - development & use" 
<live-devel at ns.live555.com>
Sent: Tuesday, May 19, 2009 2:04 AM
Subject: Re: [Live-devel] non-blocking describe in RTSPClient?


> >Is there a non-blocking Describe implementation for RTSPClient?
>
> Not yet, but for now (pending a complete reimplementation of "RTSPClient", 
> which is coming) you can use the "timeout" parameter to "describeURL()". 
> This allows you to specify an upper bound (in seconds) on how long to 
> block.
> -- 
>
> Ross Finlayson
> Live Networks, Inc.
> http://www.live555.com/
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
> 


From finlayson at live555.com  Mon May 18 18:23:08 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Mon, 18 May 2009 18:23:08 -0700
Subject: [Live-devel] NetCommon.hh and UsageEnvironment.hh
In-Reply-To: <5E7AC58DE2EC40EAAFC5263745941DB4@CSystemDev>
References: <55791956-5835-427A-86CA-6B9411F14D43@vtilt.com>
	<f06240802c637a788fe79@[66.80.62.44]>
	<5E7AC58DE2EC40EAAFC5263745941DB4@CSystemDev>
Message-ID: <f06240805c637ba245b39@[66.80.62.44]>

>    Could you confirm that NetCommon.hh is included in every cpp file 
>of Live555? And UsageEnvironment.hh?

Most, but not all.  There are a few 'pure library' files - like 
"Base64.cpp" - that don't refer to "Medium" or "UsageEnvironment" 
objects - that don't include these.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From rippeltippel at gmail.com  Tue May 19 05:43:48 2009
From: rippeltippel at gmail.com (rippel tippel)
Date: Tue, 19 May 2009 13:43:48 +0100
Subject: [Live-devel] How to force RTSP Clients disconnection
Message-ID: <c714baeb0905190543o2706a786h947cac2b0f6a1400@mail.gmail.com>

Hi, I'm working on an RTSP Server and I need to force the
disconnection of certain Clients.

>From my understanding I need to delete all the RTSPClientSession
instances, except the ones that I want to keep alive. Is it correct?

It seems that RTSPServer has no control over its RTSPClientSessions,
so I'm thinking to create a list inside the RTSPServer containing its
RTSPClientSessions, but it doesn't seem a "clean" solution.

Is there any better way to achieve that?

Regards,
R.

From vanevery at walking-productions.com  Tue May 19 12:56:25 2009
From: vanevery at walking-productions.com (Shawn Van Every)
Date: Tue, 19 May 2009 15:56:25 -0400
Subject: [Live-devel] openRTSP woes
Message-ID: <F484D9F0-724F-4152-9734-0CEE1A582FE0@walking-productions.com>

Hi Folks,

Running into a couple of issues using openRTSP and hoping for some  
advice on things that I should try or code that I should look at:

Doing recordings from Axis Q1755 H.264 HD cams using openRTSP.

Our first issue was that we would often get recordings that were  
missing a key atom and were therefore unplayable.  It was determined  
that this was due to network packet loss and some network  
reconfiguring has (mostly) solved the issue.

The second issue that we have run into is with audio/video sync.   
Right now, at the beginning of many of the recorded files, we are  
seeing very fast video with normal speed audio which I think is the  
culprit.  This lasts for less than a second.  (Using the -y flag seems  
to help but it throws away audio data making our audio very bad)  Is  
it possible this occurs until the first keyframe is seen?

We are encoding at full resolution and full framerate from these cams:

openRTSP -q -f 30 -w 1280 -h 720 -t -Q -b 2000000 rtsp://10.10.10.10/axis-media/media.amp?videocodec=h264&resolution=1280x720&audio=1&duration=0&fps=30&videobitrate=10000&videomaxbitrate=10000&videobitratepriority=framerate&videokeyframeinterval=2&compression=10&color=1&clock=0&date=0&text=0

Thanks,
shawn

From stas.oskin at gmail.com  Tue May 19 13:16:55 2009
From: stas.oskin at gmail.com (Stas Oskin)
Date: Tue, 19 May 2009 23:16:55 +0300
Subject: [Live-devel] media server capabilities
Message-ID: <77938bc20905191316g182317a1i60e4ce4bc094204a@mail.gmail.com>

Hi.

I tried to search for this info in documents and in mailing lists, but
couldn't find any.

1) Can the media server reflect RTP packets like Darwin streaming server for
example?

2) Can the media server wrap RTSP in HTTP, again, like DSS?

3) For some reason, the M4E files I create via openRTSP are not seek-able,
at least via VLC. Are there any know limitations for players trick play?

Regards.
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From finlayson at live555.com  Tue May 19 15:52:31 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Tue, 19 May 2009 15:52:31 -0700
Subject: [Live-devel] openRTSP woes
In-Reply-To: <F484D9F0-724F-4152-9734-0CEE1A582FE0@walking-productions.com>
References: <F484D9F0-724F-4152-9734-0CEE1A582FE0@walking-productions.com>
Message-ID: <f06240804c638e2a78d51@[66.80.62.44]>

>The second issue that we have run into is with audio/video sync.  
>Right now, at the beginning of many of the recorded files, we are 
>seeing very fast video with normal speed audio which I think is the 
>culprit.  This lasts for less than a second.  (Using the -y flag 
>seems to help but it throws away audio data making our audio very 
>bad)  Is it possible this occurs until the first keyframe is seen?
>
>We are encoding at full resolution and full framerate from these cams:
>
>openRTSP -q -f 30 -w 1280 -h 720 -t -Q -b 2000000 
>rtsp://10.10.10.10/axis-media/media.amp?videocodec=h264&resolution=1280x720&audio=1&duration=0&fps=30&videobitrate=10000&videomaxbitrate=10000&videobitratepriority=framerate&videokeyframeinterval=2&compression=10&color=1&clock=0&date=0&text=0


First, you should make sure that you are seeing properly synced audio 
and video when you play the stream directly, using (e.g.) VLC.  If 
(for whatever reason) you're not getting A/V sync when you play the 
stream directly, then you will never be able to get A/V sync when you 
record the data into a file.

Second, it's important to realize that the ".mov" (or ".mp4") file 
format is badly designed, and is poorly suited for recording live 
input streams (like these).  One basic problem with the file format 
is that it records audio/video data using sample/frame *durations*, 
rather than timestamps (or presentation times).  This makes it very 
difficult to keep audio/video data in sync, if the input data is 
synchronized using presentation times, as is the case for incoming 
RTP/RTCP data.  Because of this, the current implementation of 
writing ".mov" or ".mp4" files is - and will likely always remain - 
an unreliable hack.

If you want to have any chance of A/V sync working when writing 
".mov" or ".mp4" files, then you *must* use the "-y" option.  This 
may lose a small amount of initial data (until RTCP synchronization 
begins), but that's unavoidable.

If you really want to look at the appropriate code for this, then the 
class to look at is "QuickTimeFileSync".
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From finlayson at live555.com  Tue May 19 15:55:53 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Tue, 19 May 2009 15:55:53 -0700
Subject: [Live-devel] media server capabilities
In-Reply-To: <77938bc20905191316g182317a1i60e4ce4bc094204a@mail.gmail.com>
References: <77938bc20905191316g182317a1i60e4ce4bc094204a@mail.gmail.com>
Message-ID: <f06240805c638e8dc01a9@[66.80.62.44]>

>1) Can the media server reflect RTP packets like Darwin streaming 
>server for example?

No; it's a media *server*, not a 'reflector'.

>
>2) Can the media server wrap RTSP in HTTP, again, like DSS?

No, not yet, although this is a feature that we plan to support in 
the future.  (We already support if (as an option) for RTSP 
*clients*.)

>
>3) For some reason, the M4E files I create via openRTSP are not 
>seek-able, at least via VLC. Are there any know limitations for 
>players trick play?

Yes, see
	http://www.live555.com/liveMedia/faq.html#trick-mode
	http://www.live555.com/mediaServer/#trick-play

     Ross.

From ottavio at videotec.com  Wed May 20 00:06:04 2009
From: ottavio at videotec.com (Ottavio Campana)
Date: Wed, 20 May 2009 09:06:04 +0200
Subject: [Live-devel] RTP Header Extension
Message-ID: <4A13ABDC.4080402@videotec.com>

In RFC 3550 Section 5.3.1 they say that it is possible to add an header
extension that is ignored by applications that are not aware of it.

My question is: is there a way to do it with livemedia?

From seetaram.nt at gmail.com  Tue May 19 23:26:55 2009
From: seetaram.nt at gmail.com (seetaram N T)
Date: Wed, 20 May 2009 11:56:55 +0530
Subject: [Live-devel] Problem streaming *.mpeg4 file using
	testOnDemandRTSPServer.exe
Message-ID: <d77ed2070905192326i3f21982cre9e9c5b61e0b8e86@mail.gmail.com>

Hi,

I have a *.mpeg4 file which contains the mpeg4 SP encoded video. I am trying
to Stream this using testOnDemandRTSPServer.exe application on Windows PC. I
just renamed *.mpeg4 file to test.m4e and trying to stream. When i tried to
play this video using VLC, it gives the following error.

main debug: adding playlist item `rtsp://199.63.73.109:8554/mpeg4ESVideoTest'
( rtsp://199.63.73.109:8554/mpeg4ESVideoTest )

main debug: creating new input thread

main debug: waiting for thread completion

main debug: thread 4676 (input) created at priority 1 (input/input.c:265)

main debug: `rtsp://199.63.73.109:8554/mpeg4ESVideoTest' gives access `rtsp'
demux `' path `199.63.73.109:8554/mpeg4ESVideoTest'

main debug: creating demux: access='rtsp' demux='' path='
199.63.73.109:8554/mpeg4ESVideoTest'

main debug: looking for access_demux module: 1 candidate

live555 debug: RTP subsession 'video/MP4V-ES'

main debug: selecting program id=0

main debug: using access_demux module "live555"

main debug: looking for a subtitle file in D:\vlc-0.8\vlc-0.8.6i\

main debug: looking for decoder module: 28 candidates

ffmpeg debug: libavcodec already initialized

ffmpeg debug: postprocessing disabled

ffmpeg debug: using direct rendering

ffmpeg debug: ffmpeg codec (MPEG-4 Video) started

main debug: using decoder module "ffmpeg"

main debug: thread 4276 (decoder) created at priority 0
(input/decoder.c:159)

main debug: `rtsp://199.63.73.109:8554/mpeg4ESVideoTest' successfully opened

live555 debug: StreamClose

main debug: EOF reached

main debug: closing input

main debug: removing module "live555"

ffmpeg debug: ffmpeg codec (MPEG-4 Video) stopped

main debug: removing module "ffmpeg"

main debug: thread times: real 0m0.187500s, kernel 0m0.000000s, user
0m0.000000s

main debug: thread 4276 joined (input/decoder.c:191)

main debug: killing decoder fourcc `mp4v', 0 PES in FIFO

main debug: thread times: real 0m0.734375s, kernel 0m0.015625s, user
0m0.000000s

main debug: thread 4676 joined (input/input.c:412)

main: nothing to play



I read through some live555 posts and understood that, live555 doesn't
support .mpeg4 file streaming? Let me know what is the reason and what is
the solution to this problem?



thanks
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From finlayson at live555.com  Wed May 20 01:39:46 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 20 May 2009 01:39:46 -0700
Subject: [Live-devel] Problem streaming *.mpeg4 file using
 testOnDemandRTSPServer.exe
In-Reply-To: <d77ed2070905192326i3f21982cre9e9c5b61e0b8e86@mail.gmail.com>
References: <d77ed2070905192326i3f21982cre9e9c5b61e0b8e86@mail.gmail.com>
Message-ID: <f06240804c6397168a09c@[66.80.62.44]>

>I have a *.mpeg4 file which contains the mpeg4 SP encoded video. I 
>am trying to Stream this using testOnDemandRTSPServer.exe 
>application on Windows PC. I just renamed *.mpeg4 file to test.m4e
*
Where did you get the idea that this would work??  A ".mpeg4" file is 
*not* a MPEG-4 Video Elementary Stream File.  We don't currently 
support streaming from ".mpeg4" (or, equivalently, ".mp4") files.

>I read through some live555 posts and understood that, live555 
>doesn't support .mpeg4 file streaming?

Then why did you try it?

>  Let me know what is the reason

Because we haven't implemented it

>  and what is the solution to this problem?

The solution is for someone to implement it.

Geez....   Does everyone now see why I discriminate against people 
who use "@gmail.com" email addresses :-)

From finlayson at live555.com  Wed May 20 01:45:02 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 20 May 2009 01:45:02 -0700
Subject: [Live-devel] RTP Header Extension
In-Reply-To: <4A13ABDC.4080402@videotec.com>
References: <4A13ABDC.4080402@videotec.com>
Message-ID: <f06240805c63973791ca4@[66.80.62.44]>

>In RFC 3550 Section 5.3.1 they say that it is possible to add an header
>extension that is ignored by applications that are not aware of it.
>
>My question is: is there a way to do it with livemedia?

Currently not without modifying the code.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From victor at dialog.su  Wed May 20 02:59:12 2009
From: victor at dialog.su (Victor V. Vinokurov)
Date: Wed, 20 May 2009 13:59:12 +0400
Subject: [Live-devel] MPEG1or2FileServerDemux::createNew Rise exeption
Message-ID: <4A13D470.3050104@dialog.su>

i've try to get file duration of mpeg1 video file like this:

int VIDEOMC_API GetFileDuration(const char* inputFileName)
{
  TaskScheduler* scheduler = BasicTaskScheduler::createNew();
  BasicUsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);
  MPEG1or2FileServerDemux* demux =
MPEG1or2FileServerDemux::createNew(*env, inputFileName, false);
  if(demux)
    return int(demux->fileDuration());

  return 0;
}

but in VS2008 line MPEG1or2FileServerDemux::createNew rise exeption:
First-chance exception at 0x7c812afb in testvideomc.exe: Microsoft C++
exception: int at memory location 0x0011f880..
and program abnormally closed after this.

what this can mean?

-- 
See you!
---
Vityusha V. Vinokurov - programmer
mailto:victor at dialog.su
http://www.dialog.su

From ottavio at videotec.com  Wed May 20 05:16:49 2009
From: ottavio at videotec.com (Ottavio Campana)
Date: Wed, 20 May 2009 14:16:49 +0200
Subject: [Live-devel] problem with testMP3Streamer and mplayer
Message-ID: <4A13F4B1.8020004@videotec.com>

I'm trying testMP3Streamer  to multicast an MP? file over the lan and to
listen to it with mplayer. It does not work, even though it works with vlc.

Did you experience this?

Here's mplayer output

ottavio at debian2:/tmp$ mplayer -msglevel all=9
rtsp://192.168.68.3:8554/testStream
MPlayer dev-SVN-r26940
CPU: Intel(R) Core(TM)2 CPU          6600  @ 2.40GHz (Family: 6, Model:
15, Stepping: 6)
CPUflags:  MMX: 1 MMX2: 1 3DNow: 0 3DNow2: 0 SSE: 1 SSE2: 1
Compiled with runtime CPU detection.
Adding file rtsp://192.168.68.3:8554/testStream
Config pushed level is now 2
Config pushed level is now 3
get_path('codecs.conf') -> '/home/ottavio/.mplayer/codecs.conf'
Reading /home/ottavio/.mplayer/codecs.conf: Can't open
'/home/ottavio/.mplayer/codecs.conf': No such file or directory
Reading /etc/mplayer/codecs.conf: Can't open '/etc/mplayer/codecs.conf':
No such file or directory
Using built-in default codecs.conf.
Configuration: --prefix=/usr --confdir=/etc/mplayer
--datadir=/usr/share/mplayer --enable-xmga --enable-mga
--enable-joystick --enable-faad-external --disable-tremor-internal
--enable-libamr_nb --enable-libamr_wb --enable-pulse --language=all
--enable-largefiles --enable-menu --disable-libdvdcss-internal
--enable-lirc --enable-radio-capture --enable-png --enable-radio
--enable-xvmc --with-xvmclib=XvMCW --enable-fbdev --enable-tdfxfb
--enable-s3fb --enable-runtime-cpudetection --enable-gui --disable-mencoder
CommandLine: '-msglevel' 'all=9' 'rtsp://192.168.68.3:8554/testStream'
init_freetype
Using MMX (with tiny bit MMX2) Optimized OnScreenDisplay
get_path('fonts') -> '/home/ottavio/.mplayer/fonts'
Using nanosleep() timing
get_path('input.conf') -> '/home/ottavio/.mplayer/input.conf'
Can't open input config file /home/ottavio/.mplayer/input.conf: No such
file or directory
Parsing input config file /etc/mplayer/input.conf
Input config file /etc/mplayer/input.conf parsed: 89 binds
Opening joystick device /dev/input/js0
Can't open joystick device /dev/input/js0: No such file or directory
Can't init input joystick
Setting up LIRC support...
mplayer: could not connect to socket
mplayer: No such file or directory
Failed to open LIRC support. You will not be able to use your remote
control.
get_path('testStream.conf') -> '/home/ottavio/.mplayer/testStream.conf'

[[[init getch2]]]

Playing rtsp://192.168.68.3:8554/testStream.
get_path('sub/') -> '/home/ottavio/.mplayer/sub/'
STREAM_RTSP, URL: rtsp://192.168.68.3:8554/testStream
Filename for url is now rtsp://192.168.68.3:8554/testStream
Filename for url is now rtsp://192.168.68.3:8554/testStream
Resolving 192.168.68.3 for AF_INET6...
Couldn't resolve name for AF_INET6: 192.168.68.3
Connecting to server 192.168.68.3[192.168.68.3]: 8554...
SDP:
v=0
o=- 1242821667199934 1 IN IP4 192.168.68.3
s=Session streamed by "testMP3Streamer"
i=/home/ottavio/Desktop/06 Pobblemi.mp3
t=0 0
a=tool:LIVE555 Streaming Media v2009.04.20
a=type:broadcast
a=control:*
a=range:npt=0-
a=x-qt-text-nam:Session streamed by "testMP3Streamer"
a=x-qt-text-inf:/home/ottavio/Desktop/06 Pobblemi.mp3
m=audio 6666 RTP/AVP 14
c=IN IP4 239.255.42.42/1
a=control:track1

rtsp_session: unsupported RTSP server. Server type is 'unknown'.
Filename for url is now rtsp://192.168.68.3:8554/testStream
Filename for url is now rtsp://192.168.68.3:8554/testStream
STREAM_LIVE555, URL: rtsp://192.168.68.3:8554/testStream
STREAM: [RTSP and SIP] rtsp://192.168.68.3:8554/testStream
STREAM: Description: standard RTSP and SIP
STREAM: Author: Ross Finlayson
STREAM: Comment: Uses LIVE555 Streaming Media library.
s->pos=0  newpos=0  new_bufpos=0  buflen=0
Stream not seekable!
 file format detected.
Initiated "audio/MPA" RTP subsession on port 6666
==> Found audio stream: 0
No stream found.


*** uninit(0x248)
DEMUXER: freeing demuxer at 0x1e84340
DEMUXER: freeing sh_audio at 0x1e8c500

[[[uninit getch2]]]
Config poped level=2
Config poped level=1

*** uninit(0x80)
vo: x11 uninit called but X11 not initialized..

Exiting... (End of file)
max framesize was 0 bytes
ottavio at debian2:/tmp$

From yorksun at freescale.com  Wed May 20 11:45:37 2009
From: yorksun at freescale.com (York Sun)
Date: Wed, 20 May 2009 13:45:37 -0500
Subject: [Live-devel] live555.com server down?
Message-ID: <1242845137.11521.16.camel@oslab-t3>

Ross,

Is the server down? I cannot access it today.

York


From stas.oskin at gmail.com  Wed May 20 03:13:46 2009
From: stas.oskin at gmail.com (Stas Oskin)
Date: Wed, 20 May 2009 13:13:46 +0300
Subject: [Live-devel] media server capabilities
In-Reply-To: <f06240805c638e8dc01a9@66.80.62.44>
References: <77938bc20905191316g182317a1i60e4ce4bc094204a@mail.gmail.com>
	<f06240805c638e8dc01a9@66.80.62.44>
Message-ID: <77938bc20905200313i4f9fc2a5if4cd5c9ede7c955d@mail.gmail.com>

Hi.

Thanks for the replies.

No; it's a media *server*, not a 'reflector'.
>

Any idea what it would take to add such functionality? Any place I can
start?


3) For some reason, the M4E files I create via openRTSP are not seek-able,
> at least via VLC. Are there any know limitations for players trick play?
>

Yes, see
>        http://www.live555.com/liveMedia/faq.html#trick-mode
>        http://www.live555.com/mediaServer/#trick-play
>

Sorry, didn't notice the bit with seeking not supported on mpeg-4.

The steps outlined in the 1st link would be enough to add the trick play
support of MPEG-4 to mediaserver?

Regards.
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From finlayson at live555.com  Wed May 20 15:55:51 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 20 May 2009 15:55:51 -0700
Subject: [Live-devel] live555.com server down?
In-Reply-To: <1242845137.11521.16.camel@oslab-t3>
References: <1242845137.11521.16.camel@oslab-t3>
Message-ID: <f06240800c63a3a55e583@[66.80.62.44]>

>Is the server down? I cannot access it today.

Yes, our upstream network had an extended outage (from roughly 1400 
UTC - 2100 UTC on May 20th).  It's back up now.  My apologies.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From vanevery at walking-productions.com  Wed May 20 06:42:24 2009
From: vanevery at walking-productions.com (Shawn Van Every)
Date: Wed, 20 May 2009 09:42:24 -0400
Subject: [Live-devel] openRTSP woes
Message-ID: <A4B2EFAC-4B61-407B-BBA3-48CD00FFA22D@walking-productions.com>

Thanks Ross.

The audio/video are in sync when viewing the actual stream in  
QuickTime (haven't tried VLC).  Interestingly enough, we are doing  
another process which converts the RTSP stream to RTMP (Flash) and we  
see a consistent half second sync issue after that process.  The  
interesting part is that this isn't the same behavior that we see with  
the openRTSP recordings.

I hear you regarding capturing to MP4/MOV files. Do you think that  
capturing to raw files and then putting them together using another  
process would yield better results?

-s

> Message: 8
> Date: Tue, 19 May 2009 15:52:31 -0700
> From: Ross Finlayson <finlayson at live555.com>
> Subject: Re: [Live-devel] openRTSP woes
> To: LIVE555 Streaming Media - development & use
> 	<live-devel at ns.live555.com>
> Message-ID: <f06240804c638e2a78d51@[66.80.62.44]>
> Content-Type: text/plain; charset="us-ascii" ; format="flowed"
>
>> The second issue that we have run into is with audio/video sync.
>> Right now, at the beginning of many of the recorded files, we are
>> seeing very fast video with normal speed audio which I think is the
>> culprit.  This lasts for less than a second.  (Using the -y flag
>> seems to help but it throws away audio data making our audio very
>> bad)  Is it possible this occurs until the first keyframe is seen?
>>
>> We are encoding at full resolution and full framerate from these  
>> cams:
>>
>> openRTSP -q -f 30 -w 1280 -h 720 -t -Q -b 2000000
>> rtsp://10.10.10.10/axis-media/media.amp?videocodec=h264&resolution=1280x720&audio=1&duration=0&fps=30&videobitrate=10000&videomaxbitrate=10000&videobitratepriority=framerate&videokeyframeinterval=2&compression=10&color=1&clock=0&date=0&text=0
>
>
> First, you should make sure that you are seeing properly synced audio
> and video when you play the stream directly, using (e.g.) VLC.  If
> (for whatever reason) you're not getting A/V sync when you play the
> stream directly, then you will never be able to get A/V sync when you
> record the data into a file.
>
> Second, it's important to realize that the ".mov" (or ".mp4") file
> format is badly designed, and is poorly suited for recording live
> input streams (like these).  One basic problem with the file format
> is that it records audio/video data using sample/frame *durations*,
> rather than timestamps (or presentation times).  This makes it very
> difficult to keep audio/video data in sync, if the input data is
> synchronized using presentation times, as is the case for incoming
> RTP/RTCP data.  Because of this, the current implementation of
> writing ".mov" or ".mp4" files is - and will likely always remain -
> an unreliable hack.
>
> If you want to have any chance of A/V sync working when writing
> ".mov" or ".mp4" files, then you *must* use the "-y" option.  This
> may lose a small amount of initial data (until RTCP synchronization
> begins), but that's unavoidable.
>
> If you really want to look at the appropriate code for this, then the
> class to look at is "QuickTimeFileSync".
> -- 
>
> Ross Finlayson
> Live Networks, Inc.
> http://www.live555.com/

From finlayson at live555.com  Wed May 20 17:51:15 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 20 May 2009 17:51:15 -0700
Subject: [Live-devel] media server capabilities
In-Reply-To: <77938bc20905200313i4f9fc2a5if4cd5c9ede7c955d@mail.gmail.com>
References: <77938bc20905191316g182317a1i60e4ce4bc094204a@mail.gmail.com>
	<f06240805c638e8dc01a9@66.80.62.44>
	<77938bc20905200313i4f9fc2a5if4cd5c9ede7c955d@mail.gmail.com>
Message-ID: <f06240800c63a55f45ee2@[66.80.62.44]>

>3) For some reason, the M4E files I create via openRTSP are not 
>seek-able, at least via VLC. Are there any know limitations for 
>players trick play?
>
>
>Yes, see
> 
> <http://www.live555.com/liveMedia/faq.html#trick-mode>http://www.live555.com/liveMedia/faq.html#trick-mode
> 
> <http://www.live555.com/mediaServer/#trick-play>http://www.live555.com/mediaServer/#trick-play
>
>
>Sorry, didn't notice the bit with seeking not supported on mpeg-4.
>
>The steps outlined in the 1st link would be enough to add the trick 
>play support of MPEG-4 to mediaserver?

Yes.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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From finlayson at live555.com  Wed May 20 17:58:37 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 20 May 2009 17:58:37 -0700
Subject: [Live-devel] MPEG1or2FileServerDemux::createNew Rise exeption
In-Reply-To: <4A13D470.3050104@dialog.su>
References: <4A13D470.3050104@dialog.su>
Message-ID: <f06240801c63a570ea12a@[66.80.62.44]>

>but in VS2008 line MPEG1or2FileServerDemux::createNew rise exeption:
>First-chance exception at 0x7c812afb in testvideomc.exe: Microsoft C++
>exception: int at memory location 0x0011f880..
>and program abnormally closed after this.
>
>what this can mean?

It means that a memory access exception occurred somewhere in the 
code, most likely somewhere in the code for 
"MPEG1or2ProgramStreamFileDuration()" (see 
"liveMedia/MPEG1or2FileServerDemux.cpp").  Fortunately, You Have 
Complete Source Code, which should help you find the problem.

(Note that "MPEG1or2FileServerDemux" assumes that its input file is a 
MPEG-1 or 2 *Program Stream" file.  If this is not the case, then 
that might have caused the code to fail.)
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From finlayson at live555.com  Wed May 20 18:20:52 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 20 May 2009 18:20:52 -0700
Subject: [Live-devel] openRTSP woes
In-Reply-To: <A4B2EFAC-4B61-407B-BBA3-48CD00FFA22D@walking-productions.com>
References: <A4B2EFAC-4B61-407B-BBA3-48CD00FFA22D@walking-productions.com>
Message-ID: <f06240800c63a5c37d69c@[66.80.62.44]>

>I hear you regarding capturing to MP4/MOV files. Do you think that 
>capturing to raw files and then putting them together using another 
>process would yield better results?

Unfortunately not, because the 'raw' (i.e., audio and video 
Elementary Stream) files would not contain any of the 'presentation 
time' information that we get from RTP/RTCP.  Therefore, when you 
later combine them into a single audio+video file, you won't be able 
to get A/V synchronization.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From finlayson at live555.com  Wed May 20 18:23:30 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 20 May 2009 18:23:30 -0700
Subject: [Live-devel] problem with testMP3Streamer and mplayer
In-Reply-To: <4A13F4B1.8020004@videotec.com>
References: <4A13F4B1.8020004@videotec.com>
Message-ID: <f06240801c63a5d311157@[66.80.62.44]>

>I'm trying testMP3Streamer  to multicast an MP? file over the lan and to
>listen to it with mplayer. It does not work, even though it works with vlc.

Because this seems to be a MPlayer-specific 
problem, you should ask about it on a MPlayer 
mailing list.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From workzh at hotmail.com  Wed May 20 18:24:44 2009
From: workzh at hotmail.com (zhBruce)
Date: Thu, 21 May 2009 01:24:44 +0000
Subject: [Live-devel] Question about "MP4A-LATM" and "mpeg4-generic".
Message-ID: <BLU139-W9568358F5DEAF2F560F41DC590@phx.gbl>


Hi all:
   
 I'm developing a rtp receiving directshow filter, using live555
as the rtp/rtsp module. I adopt the Apple's Darwin streaming server.
At first, I converted a media file to 3gp format named as mov1.3gp 
using QuickTime Pro, with video format is H.263 and audio is AAC-LC.
Then I can receive the stream and play it well.
 

Yesterday, I converted a media file using ffmpeg,not Quick
Time Pro,I selected the H263 video format and AAC(using option:
-acodec libfaac),I put the file on server to steam it. On the client
I can receive both video and audio RTP packets,But can only play video.
I checked the SDP data from server,found that when server streams 
media file generated by QT Pro,the audio is "MP4A-LATM",and when 
streams file generated by ffmpeg,audio is "mpeg4-generic".


 The first 3gp file was converted and hinted by QT Pro,The 
Second was converted by ffmpeg and hinted by mp4creator.

 

 I can make sure that both media files have the same AAC
audio format. Someone could tell me why the same format ,but different
way server streams them? And , the other question is ,what exectly
is the difference between "MP4A-LATM" and "mpeg4-generic".

  Thanks in advance..

                                                                 

                                                                         Bruce

_________________________________________________________________
????Live????Windows Live????, ????????
http://events.livetome.cn/2009/knowlive
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From finlayson at live555.com  Wed May 20 19:12:24 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 20 May 2009 19:12:24 -0700
Subject: [Live-devel] Question about "MP4A-LATM" and "mpeg4-generic".
In-Reply-To: <BLU139-W9568358F5DEAF2F560F41DC590@phx.gbl>
References: <BLU139-W9568358F5DEAF2F560F41DC590@phx.gbl>
Message-ID: <f06240803c63a67a08348@[66.80.62.44]>

>Someone could tell me why the same format ,but different
>way server streams them?

Because different organizations (perhaps for political reasons, I'm 
not sure) came up with different RTP payload formats for streaming 
the same type of data (MPEG-4 audio).


>  And , the other question is ,what exectly
>is the difference between "MP4A-LATM" and "mpeg4-generic".

They're different RTP payload formats.  The former is described in 
RFC 3016 (and implemented by us in the "MPEG4LATMAudioRTPSink" and 
"MPEG4LATMAudioRTPSource" classes); the latter is described in RFC 
3640 (and implemented by us in the "MPEG4GenericRTPSink" and 
"MPEG4GenericRTPSource" classes).
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From tadmorm1 at post.tau.ac.il  Thu May 21 06:19:14 2009
From: tadmorm1 at post.tau.ac.il (tadmorm1 at post.tau.ac.il)
Date: Thu, 21 May 2009 16:19:14 +0300
Subject: [Live-devel] Packet Loss From VLS
Message-ID: <20090521161914.42144tp5iyyei95e@webmail.tau.ac.il>

Hi,
I'm using the simpleRTPsource in order to receive rtp/udp packets  
which encapsulates TS (encapsulating an h.264 video stream). only made  
a simple adjustment to testMPEG1or2VideoReceiver.

When my source is Live555 and the testMPEG2TransportStreamer it's fine.
However, I'm experiencing heavy packet loss when I transmit from the  
vls program. I tried increasing the internal buffers using  
"increaseReceiveBufferTo" as suggested in the FAQ , but it didn't help.
Any ideas why this happens? and what can I do?
can it be related to different bitrate of the streaming programs?
Thanks for the help,
Michael




From finlayson at live555.com  Thu May 21 07:34:33 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Thu, 21 May 2009 07:34:33 -0700
Subject: [Live-devel] Packet Loss From VLS
In-Reply-To: <20090521161914.42144tp5iyyei95e@webmail.tau.ac.il>
References: <20090521161914.42144tp5iyyei95e@webmail.tau.ac.il>
Message-ID: <f06240806c63b16257734@[66.80.62.44]>

>I'm using the simpleRTPsource in order to receive rtp/udp packets 
>which encapsulates TS (encapsulating an h.264 video stream). only 
>made a simple adjustment to testMPEG1or2VideoReceiver.
>
>When my source is Live555 and the testMPEG2TransportStreamer it's fine.
>However, I'm experiencing heavy packet loss when I transmit from the 
>vls program

Then the problem is with that program.  You'll need to ask the 
authors of that program what's wrong.

(I suspect, though, that they might be trying to send one 188-byte 
Transport Stream packet per RTP/UDP packet, instead of aggregating 
together 7 Transport Stream packets into each RTP/UDP packet (which 
is what most people do).  But I'm just speculating.  VLS is not our 
software.  Bug reports on that software are off-topic for this 
mailing list.)
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From ganesh_vijayan at yahoo.com  Thu May 21 09:35:04 2009
From: ganesh_vijayan at yahoo.com (Ganesh V)
Date: Thu, 21 May 2009 09:35:04 -0700 (PDT)
Subject: [Live-devel] Doubt on multiple media subsessions
Message-ID: <800241.23315.qm@web39502.mail.mud.yahoo.com>

Dear Experts,

I am trying to decode an incoming stream over RTSP. When I checked the SDP description, I observed that there are 2 media subsessions like H.264 and G.711. To decode the same, I am trying to connect 2 separate sinks for each of the individual sub-sessions.

Is my idea of implementation correct? Should I perform any other initialization during SDP initialization, especially given that one is a video and another an audio session?

Thanks,
Ganesh



      
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From finlayson at live555.com  Thu May 21 13:25:33 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Thu, 21 May 2009 13:25:33 -0700
Subject: [Live-devel] Doubt on multiple media subsessions
In-Reply-To: <800241.23315.qm@web39502.mail.mud.yahoo.com>
References: <800241.23315.qm@web39502.mail.mud.yahoo.com>
Message-ID: <f06240808c63b683ee91b@[66.80.62.44]>

>I am trying to decode an incoming stream over RTSP. When I checked 
>the SDP description, I observed that there are 2 media subsessions 
>like H.264 and G.711. To decode the same, I am trying to connect 2 
>separate sinks for each of the individual sub-sessions.
>
>Is my idea of implementation correct?

Yes.  Note that this is just what the "openRTSP" code does (see 
"testProgs/playCommon.cpp").

Just make sure that each of your 'sink' objects uses the input 
"presentationTime" value to properly synchronize the streams.

>  Should I perform any other initialization during SDP 
>initialization, especially given that one is a video and another an 
>audio session?

No, calling "MediaSubsession::initiate()" on each subsession is 
sufficient.  Our code automatically figures out what kind of 
"RTPSource" object to create, based on the SDP description.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From igor.milavec at lsi.si  Fri May 22 01:53:46 2009
From: igor.milavec at lsi.si (Igor Milavec)
Date: Fri, 22 May 2009 10:53:46 +0200
Subject: [Live-devel] exit()
Message-ID: <A2518FB4852F974483E88CD74D74B78701D7463CBD5F@perun.l-sol.si>

Hi.

I had a >little< problem with my application just closing for no apparent reason. Then I realized that LiveMedia library calls exit() in case of errors... I'm developing an application that displays multiple video streams and if there is nomething wrong with one of the streams I definitely do not want for the application to just die on me...

That's why I propose to either:

-          Use C++ exceptions where possible

-          add the exit() virtual method to the Environment which in turn will call standard library exit() so the user can override it

Regards, Igor
-----
Igor Milavec
Li?er Solutions d.o.o.
Cesta Andreja Bitenca 68
SI-1000 Ljubljana
Tel:        +386 1 5101-780
Fax:       +386 1 5101-785

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From finlayson at live555.com  Fri May 22 02:18:30 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Fri, 22 May 2009 02:18:30 -0700
Subject: [Live-devel] exit()
In-Reply-To: <A2518FB4852F974483E88CD74D74B78701D7463CBD5F@perun.l-sol.si>
References: <A2518FB4852F974483E88CD74D74B78701D7463CBD5F@perun.l-sol.si>
Message-ID: <f0624080cc63c1e219578@[66.80.62.44]>

>I had a ?little? problem with my application 
>just closing for no apparent reason. Then I 
>realized that LiveMedia library calls exit() in 
>case of errors...

This happens in only a few, special places in the 
code, and usually indicates a serious, 
unrecoverably bug somewhere.

Which specific instance of "exit()" in the code is causing you problems?
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From herlit11 at lycos.com  Fri May 22 18:33:46 2009
From: herlit11 at lycos.com (her Garcia)
Date: Fri, 22 May 2009 21:33:46 -0400 (EDT)
Subject: [Live-devel] 404 Stream not  found
Message-ID: <20090522213346.HM.0000000000003zf@herlit11.mail-wwl21.bo3.lycos.com.lycos.com>

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Hello, everyone. I am new to live555 and i am having a problema streaming.

I started the server to stream an mpeg2video file and when i try to get it with
vlc or my set top box(istar mini hd) i get the error:

404 Stream not found.

I tried to sniff the rtsp stream with wireshark and i found that the server 
response to my set top box is changed,the url or stream name changes
from the following:

rtsp://10.0.0.3/capture-live.mpg 
to 
rtsp://10.0.0.3/capture-live.mpg/

Please notice the extra "/" at the ending. I think the server response is changing
what is suppossed to be a file into a directory and it's messing up my set top box
connection.

Plase see the following rtsp conversation from wireshark:


The client:

		
	DESCRIBE rtsp://10.0.0.3/capture-live.mpg RTSP/1.0


CSeq: 1


Accept: application/sdp


User-Agent: WMPlayer/9.0.0.3250 guid/3300AD50-2C39-46C0-AE0A-27CFF7344562



The server response:

RTSP/1.0 200 OK


CSeq: 1


Date: Fri, May 15 2009 13:56:46 GMT


Content-Base: rtsp://10.0.0.3/capture-live.mpg/


Content-Type: application/sdp


Content-Length: 489


Is the Content-Base setting of the rtsp/sdp changing the stream name, thus creating
the error and the following "404 Stream not Found" ?


Any ideas, thanks in advance,sorry for the long mail,regards
Herlit11


From finlayson at live555.com  Fri May 22 23:15:59 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Fri, 22 May 2009 23:15:59 -0700
Subject: [Live-devel] 404 Stream not  found
In-Reply-To: <20090522213346.HM.0000000000003zf@herlit11.mail-wwl21.bo3.lycos.com.lycos
	.com>
References: <20090522213346.HM.0000000000003zf@herlit11.mail-wwl21.bo3.lycos.com.lycos
	.com>
Message-ID: <f06240802c63d446fca6f@[66.80.62.44]>

>The server response:
>RTSP/1.0 200 OK
>
>CSeq: 1
>
>Date: Fri, May 15 2009 13:56:46 GMT
>
>Content-Base: rtsp://10.0.0.3/capture-live.mpg/

This is actually not an error, because the client should then be 
asking to "PLAY" each specific track, using a track id that it will 
append to the "Content-Base:" URL.

What happens when you try playing the stream using "openRTSP"?  Note 
the RTSP protocol exchange that "openRTSP" outputs.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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From herlit11 at lycos.com  Sat May 23 05:47:36 2009
From: herlit11 at lycos.com (her Garcia)
Date: Sat, 23 May 2009 08:47:36 -0400 (EDT)
Subject: [Live-devel] 404 Stream not found
Message-ID: <20090523084736.HM.00000000000040X@herlit11.mail-wwl21.bo3.lycos.com.lycos.com>

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>This is actually not an error, because the client should then be 
>asking to "PLAY" each specific track, using a track id that it will 
>append to the "Content-Base:" URL.

>What happens when you try playing the stream using "openRTSP"? Note 
>the RTSP protocol exchange that "openRTSP" outputs.

Thanks for your reply:
 
I see that the server presents the track1 and track2 and then the client does the SETUP,but there is no
PLAY of each specific track. I will try with the openRTSP to see what's going on. 

Thanks again,regards
Hernan

From finlayson at live555.com  Sat May 23 07:26:19 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Sat, 23 May 2009 07:26:19 -0700
Subject: [Live-devel] 404 Stream not found
In-Reply-To: <20090523084736.HM.00000000000040X@herlit11.mail-wwl21.bo3.lycos.com.lycos
	.com>
References: <20090523084736.HM.00000000000040X@herlit11.mail-wwl21.bo3.lycos.com.lycos
	.com>
Message-ID: <f06240800c63db75596c9@[66.80.62.44]>

>I see that the server presents the track1 and track2 and then the 
>client does the SETUP,but there is no
>PLAY of each specific track.

Does this happen even when VLC is used as the client??  VLC uses our 
RTSP client implementation, so it should be working OK

>  I will try with the openRTSP to see what's going on.

Please do.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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From herlit11 at lycos.com  Sat May 23 09:18:17 2009
From: herlit11 at lycos.com (her Garcia)
Date: Sat, 23 May 2009 12:18:17 -0400 (EDT)
Subject: [Live-devel] 404 Stream not found
Message-ID: <20090523121817.HM.00000000000040v@herlit11.mail-wwl21.bo3.lycos.com.lycos.com>

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I did nt try with vlc. Rigth now i am trying to duplicate the tests on my notebook at home.
I have a different error now(on my notebook installation) so i cant run the same tests  yet.

 Should i send another mail to the list with thie new issue to avoid top posting?



 Regards,thanks
Hernan








---------[ Received Mail Content ]----------

 Subject : Re: [Live-devel] 404 Stream not found

 Date : Sat, 23 May 2009 07:26:19 -0700

 From : Ross Finlayson <finlayson at live555.com>

 To : LIVE555 Streaming Media - development &amp; use 	<live-devel at ns.live555.com>



>I see that the server presents the track1 and track2 and then the 

>client does the SETUP,but there is no

>PLAY of each specific track.



Does this happen even when VLC is used as the client?? VLC uses our 

RTSP client implementation, so it should be working OK



> I will try with the openRTSP to see what's going on.



Please do.

-- 



Ross Finlayson

Live Networks, Inc.

http://www.live555.com/

From finlayson at live555.com  Sat May 23 11:44:42 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Sat, 23 May 2009 11:44:42 -0700
Subject: [Live-devel] 404 Stream not found
In-Reply-To: <20090523121817.HM.00000000000040v@herlit11.mail-wwl21.bo3.lycos.com.lycos
	.com>
References: <20090523121817.HM.00000000000040v@herlit11.mail-wwl21.bo3.lycos.com.lycos
	.com>
Message-ID: <f06240801c63df4077897@[66.80.62.44]>

>p {margin-top:0px;margin-bottom:0px;}
>I did nt try with vlc. Rigth now i am trying to duplicate the tests 
>on my notebook at home.
>I have a different error now(on my notebook installation) so i cant 
>run the same tests  yet.
>
>  Should i send another mail to the list with thie new issue to avoid 
>top posting?

I'm interested in seeing a bug report *only* if you can reproduce it 
with our original, unmodified server code - either 
"testMPEG1or2AudioVideoStreamer", "testOnDemandRTSPServer", or 
"live555MediaServer" - and using "openRTSP" as a client.  If you 
can't reproduce the bug using our original, unmodified software, then 
I'm not interested.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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From ben at decadent.org.uk  Sun May 24 08:15:28 2009
From: ben at decadent.org.uk (Ben Hutchings)
Date: Sun, 24 May 2009 16:15:28 +0100
Subject: [Live-devel] [PATCH] Add SMPTE 370M (DVCPRO HD) profiles
Message-ID: <1243178128.16597.72.camel@deadeye>

This changes the DV profile matching to match ffmpeg's code, adding
support for DVCPRO HD.  I have no examples of DVPCRO HD streams to test
against, but presumably ffmpeg does the right thing.

Ben.

diff --git a/liveMedia/DVVideo.cpp b/liveMedia/DVVideo.cpp
index 4b8742f..3dc3bc9 100644
--- a/liveMedia/DVVideo.cpp
+++ b/liveMedia/DVVideo.cpp
@@ -20,26 +20,23 @@ along with this library; if not, write to the Free Software Foundation, Inc.,
 #include <stddef.h>
 #include "DVVideoInternals.hh"
 
-// We support the following profiles:
-//
-// SD-VCR/525-60
-// SD-VCR/625-50
-// 314M-25/525-60
-// 314M-25/625-50
-// 314M-50/525-60
-// 314M-50/625-50
-//
-// We don't include HD-VCR (MPEG-2 payload), SDL-VCR (never seen it),
-// or 370M (never seen it).  RFC 3189bis says 306M can be treated as
-// 314M-25.
+// The profiles here do not include HD-VCR (this has an MPEG-2 payload
+// and therefore entirely different presentation timing) or SDL-VCR
+// (the author has never seen any specification or codec for it).
+// SMPTE 314M obsoleted 306M and there is no need to treat them as
+// defining separate profiles.
 
 static DVVideoProfile const profiles[] = {
-  { "SD-VCR/525-60",  0, 10, 1, 1000000 * 1001 / 30000 },
-  { "SD-VCR/625-50",  0, 12, 1, 1000000 / 25 },
-  { "314M-25/525-60", 1, 10, 1, 1000000 * 1001 / 30000 },
-  { "314M-25/625-50", 1, 12, 1, 1000000 / 25 },
-  { "314M-50/525-60", 1, 10, 2, 1000000 * 1001 / 30000 },
-  { "314M-50/625-50", 1, 12, 2, 1000000 / 25 },
+  { "SD-VCR/525-60",  0, 0x00, 10, 1, 1000000 * 1001 / 30000 },
+  { "SD-VCR/625-50",  0, 0x00, 12, 1, 1000000 / 25 },
+  { "314M-25/525-60", 1, 0x00, 10, 1, 1000000 * 1001 / 30000 },
+  { "314M-25/625-50", 1, 0x00, 12, 1, 1000000 / 25 },
+  { "314M-50/525-60", 1, 0x04, 10, 2, 1000000 * 1001 / 30000 },
+  { "314M-50/625-50", 1, 0x04, 12, 2, 1000000 / 25 },
+  { "370M/1080-60i",  1, 0x14, 10, 4, 1000000 * 1001 / 30000 },
+  { "370M/1080-50i",  1, 0x14, 12, 4, 1000000 / 25 },
+  { "370M/720-60p",   1, 0x18, 10, 4, 1000000 * 1001 / 60000 },
+  { "370M/720-50p",   1, 0x18, 12, 4, 1000000 / 50 },
   { 0 }
 };
 
@@ -53,18 +50,16 @@ const DVVideoProfile* DVVideoProfile::getByHeader(const DVBlock* headerBlocks)
     return NULL;
 
   unsigned apt = headerBlocks[0].Data[1] & 7;
+  unsigned stype = headerBlocks[5].Data[48] & 0x1f;
   unsigned sequenceCount =
     (headerBlocks[0].Data[0] == DVPackHeader12) ? 12 : 10;
-  unsigned channelCount =
-    (headerBlocks[5].Data[45] == DVPackVideoSource &&
-     headerBlocks[5].Data[48] & 4) ? 2 : 1;
 
   for (DVVideoProfile const* profile = profiles;
        profile->Name != NULL;
        ++profile)
     if (profile->APT == apt &&
-	profile->SequenceCount == sequenceCount &&
-	profile->ChannelCount == channelCount)
+	profile->SType == stype &&
+	profile->SequenceCount == sequenceCount)
       return profile;
 
   return NULL;
diff --git a/liveMedia/DVVideoInternals.hh b/liveMedia/DVVideoInternals.hh
index b204c56..d6f5222 100644
--- a/liveMedia/DVVideoInternals.hh
+++ b/liveMedia/DVVideoInternals.hh
@@ -91,6 +91,7 @@ struct DVVideoProfile {
 
   char const* Name;
   unsigned APT;
+  unsigned SType;
   unsigned SequenceCount;
   unsigned ChannelCount; /* for higher bit rates */
   unsigned FrameDuration; /* in microseconds */

-- 
Ben Hutchings
Teamwork is essential - it allows you to blame someone else.
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From stas.oskin at gmail.com  Mon May 25 01:57:30 2009
From: stas.oskin at gmail.com (Stas Oskin)
Date: Mon, 25 May 2009 11:57:30 +0300
Subject: [Live-devel] openRTSP woes
In-Reply-To: <f06240800c63a5c37d69c@66.80.62.44>
References: <A4B2EFAC-4B61-407B-BBA3-48CD00FFA22D@walking-productions.com>
	<f06240800c63a5c37d69c@66.80.62.44>
Message-ID: <77938bc20905250157x241b9d00q937b51d9b677ded3@mail.gmail.com>

Hi.

Unfortunately not, because the 'raw' (i.e., audio and video Elementary
> Stream) files would not contain any of the 'presentation time' information
> that we get from RTP/RTCP.  Therefore, when you later combine them into a
> single audio+video file, you won't be able to get A/V synchronization.
>

I'd like to tune in here, because it something I work on as well.

1) Just to verify, does it means that elementary stream files (m4e or m4v)
are unable to contain the PTS/DTS timestamps at all?

2) If yes, what would be the recommended format to contain the streaming
media, taking into account the variating frame-rate coming with it?

3) Perhaps it's possible to time the frames according to 90000 value and
then specify their timestamps accordingly (multiple the millisecond
timestamps by 90 for example)?

Thanks.
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From stas.oskin at gmail.com  Mon May 25 10:34:02 2009
From: stas.oskin at gmail.com (Stas Oskin)
Date: Mon, 25 May 2009 20:34:02 +0300
Subject: [Live-devel] openRTSP woes
In-Reply-To: <f06240804c638e2a78d51@66.80.62.44>
References: <F484D9F0-724F-4152-9734-0CEE1A582FE0@walking-productions.com>
	<f06240804c638e2a78d51@66.80.62.44>
Message-ID: <77938bc20905251034o30732c05v9d18deaa7c1e077b@mail.gmail.com>

Hi.

Second, it's important to realize that the ".mov" (or ".mp4") file format is
> badly designed, and is poorly suited for recording live input streams (like
> these).  One basic problem with the file format is that it records
> audio/video data using sample/frame *durations*, rather than timestamps (or
> presentation times).
>


I'd also appreciate if someone can provide a short explanation, or point me
toward an article explaining this difference between mp4 and other formats
(m4e/m4v?).

Thanks in advance.
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From alexone1980 at yahoo.it  Mon May 25 16:04:35 2009
From: alexone1980 at yahoo.it (Alex Solbi)
Date: Mon, 25 May 2009 23:04:35 +0000 (GMT)
Subject: [Live-devel] Two quick questions ( 1) OutputPacketBuffer and
	fDurationInMicroseconds 2) skipping frames in doGetNextFrame() )
Message-ID: <833439.49779.qm@web23503.mail.ird.yahoo.com>

Hi

1) Suppose that I set OutputPacketBuffer::maxSize to a large value.

Now, I wonder:

is fDurationInMicroseconds, inside doGetNextFrame() of a MySource, ignored until the buffer is filled (so it has no effect if I set it to a value different than 0) ?

---------------------------------------------------------------------------------

2)

which is the best way to "skip" frames, when calling doGetNextFrame() ? I wonder if it consists in setting fFrameSize = 0 (if so, what about the other members? fTo, fDurationInMicroseconds, fPresentationTime...)

Thanks

Alessandro





      
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From workzh at hotmail.com  Mon May 25 18:47:15 2009
From: workzh at hotmail.com (zhBruce)
Date: Tue, 26 May 2009 01:47:15 +0000
Subject: [Live-devel] Memory leaks, only creatNew But no delete
Message-ID: <BLU139-W3C3621C094128B71DA20EDC520@phx.gbl>


Hello:

 

 At first, Thanks to Ross for help on "mepg4-latm".

 Here is another trouble, I made a vs2005 project,which include
some files of live555. In the main() function in file main.c , I copied 
code in main() function in playCommon.c of live555. Building,Running 

successfully. The program receive streams and save data
in file. But at last I found that there is an important issue: memory 

meak.
 
A memory leak detection tool called "vld" output these messages when I 

debug:
-------------------------------------------
BasicTaskScheduler0::scheduleDelayedTask
...
basicusageenvironment.cpp (48): BasicUsageEnvironment::createNew
...
basictaskscheduler0.cpp (51): BasicTaskScheduler0::BasicTaskScheduler0
...
basictaskscheduler.cpp (34): BasicTaskScheduler::createNew
...
-------------------------------------------
So there exists 4 memory leaks .

I searched all files of live555, found nowhere to delete the pointer of BasicTaskScheduler instance, include playcommon.c . I don't know whether I should delete it myself.
But the destructor of class "BasicUsageEnvironment" is declared in
protected field and cannot accessed by "delete". Should I move it to
public field? Actually I've tried but memory leaks still exist.


_________________________________________________________________
Messenger??????????????????Messenger???
http://im.live.cn/safe/
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From joeflin at 126.com  Tue May 26 17:50:53 2009
From: joeflin at 126.com (joeflin)
Date: Wed, 27 May 2009 08:50:53 +0800 (CST)
Subject: [Live-devel] live555 bitrate
Message-ID: <9458612.40691243385453879.JavaMail.coremail@bj126app26.126.com>

Hi, I am using live555 to stream 264 video over the net. 

>From VLC, it shows that the bitrate of the received stream is, say, from 1386 kb/s to 2123 kb/s, even I set the bitrate actually to 5000 kb/s. 

If I dump the frames to a local file before stream it, then VLC can display this file OK at 5000 kb/s. 

So apparently, live555 streaming is limiting the bitrate. (???), or it is instructed to do so (???).

So, I launched wireshark, the trace shows that the time between 2 frames is about 0.0005 seconds. This means, 2000 "sends" per second as a constant? This number * (whatever bytes (upto MTU??) in in each "send") = total bitrate ?? Is this calculation right?
What could effect the 2000?

I tried to change the fDurationInMicroseconds, the result stays the same....

Thanks a ton!

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From finlayson at live555.com  Tue May 26 17:45:15 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Tue, 26 May 2009 17:45:15 -0700
Subject: [Live-devel] Memory leaks, only creatNew But no delete
In-Reply-To: <BLU139-W3C3621C094128B71DA20EDC520@phx.gbl>
References: <BLU139-W3C3621C094128B71DA20EDC520@phx.gbl>
Message-ID: <f06240800c6423c097101@[66.80.62.44]>

>I searched all files of live555, found nowhere to delete the pointer 
>of BasicTaskScheduler instance, include playcommon.c . I don't know 
>whether I should delete it myself.
>But the destructor of class "BasicUsageEnvironment" is declared in
>protected field and cannot accessed by "delete". Should I move it to
>public field?

No, you do not need to modfify the existing code.

To delete a UsageEnvironment* "env", call
	env->reclaim();

To delete a TaskScheduler* "sched", call
	delete sched;

You should do this in the order listed above - i.e, delete the 
UsageEnvironment first, then the TaskScheduler.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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From thomas at drewermann.org  Mon May 25 17:38:09 2009
From: thomas at drewermann.org (Thomas Drewermann)
Date: Tue, 26 May 2009 02:38:09 +0200
Subject: [Live-devel] RTSPClient as MediaSource for Live555 Media Server
Message-ID: <4A1B561102000055000020B7@mail.wg1337.de>

Hi,
 
first of all thanks for your wonderful library with so many features!
 
I'm currently faced to a problem caused by VLC.
The current VLC implementation does not support RTP over TCP to pass Client-Sided NAT environments.
 
I've tried several Proxy-Servers to pass the incoming UDP-traffic to the client via TCP but don't succeed.
 
After I spend the whole weekend trying to get VLC-stream to work with my natted client I've decided to look for a easyier solution.
I'm going to implement a RTSP-Client-Class as MediaSource for the existing Live555 Media Server.
This would bypass the problem of the client behind a nat environment because Live555 Media Server does automatic TCP-fallback when acting as RTSP-Server.
 
Is anybody out there who's done that before?
If this is the case it would be nice if you can provide me some source-code.
Does anyone has some useful hints or advices in this background?
 
Thanks in advance
Thomas
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From finlayson at live555.com  Tue May 26 18:05:14 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Tue, 26 May 2009 18:05:14 -0700
Subject: [Live-devel] live555 bitrate
In-Reply-To: <9458612.40691243385453879.JavaMail.coremail@bj126app26.126.com>
References: <9458612.40691243385453879.JavaMail.coremail@bj126app26.126.com>
Message-ID: <f06240801c64241e89dd3@[66.80.62.44]>

The rate at which data is sent through a "RTPSink" is determined 
*entirely* by the value of "fPresentationTime" that is set for each 
chunk of data that is fed to the "RTPSink".

If you are seeing a different data rate at the receiving end, it 
might be because of network packet loss.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From finlayson at live555.com  Wed May 27 01:32:28 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 27 May 2009 01:32:28 -0700
Subject: [Live-devel] openRTSP woes
In-Reply-To: <77938bc20905250157x241b9d00q937b51d9b677ded3@mail.gmail.com>
References: <A4B2EFAC-4B61-407B-BBA3-48CD00FFA22D@walking-productions.com>
	<f06240800c63a5c37d69c@66.80.62.44>
	<77938bc20905250157x241b9d00q937b51d9b677ded3@mail.gmail.com>
Message-ID: <f06240803c642aa77f3db@[66.80.62.44]>

>1) Just to verify, does it means that elementary stream files (m4e 
>or m4v) are unable to contain the PTS/DTS timestamps at all?

The video elementary stream files might contain these timestamps; the 
audio elementary files typically won't.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From finlayson at live555.com  Wed May 27 01:56:16 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 27 May 2009 01:56:16 -0700
Subject: [Live-devel] Two quick questions ( 1) OutputPacketBuffer and
 fDurationInMicroseconds 2) skipping frames in doGetNextFrame() )
In-Reply-To: <833439.49779.qm@web23503.mail.ird.yahoo.com>
References: <833439.49779.qm@web23503.mail.ird.yahoo.com>
Message-ID: <f06240805c642ae40d713@[66.80.62.44]>

>is fDurationInMicroseconds, inside doGetNextFrame() of a MySource, 
>ignored until the buffer is filled (so it has no effect if I set it 
>to a value different than 0) ?

It (and all the other parameters) are ignored until 
"FramedSource::afterGetting()" is called.


>which is the best way to "skip" frames, when calling doGetNextFrame() ?

Just don't call "FramedSource::afterGetting()" for frames that you 
want to skip.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From zmax.linkedin at gmail.com  Wed May 27 06:39:11 2009
From: zmax.linkedin at gmail.com (Massimo Zito)
Date: Wed, 27 May 2009 15:39:11 +0200
Subject: [Live-devel] H264 TS
Message-ID: <92a42b330905270639mb3cb88dx13ef1473c2af1b90@mail.gmail.com>

Hi Ross,

in attach my modified transport stream multiplexor to handle h264 stream
type ...

It works for me ... Is this correct ?

Thanks ...

Massimo


*** MPEG2TransportStreamMultiplexor.cpp 2009-04-07 04:19:00.000000000 +0200
--- MPEG2TransportStreamMultiplexor.cpp.new     2009-05-27
15:24:43.000000000 +0200
***************
*** 109,115 ****
        // Instead, set the stream's type to default values, based on
whether
        // the stream is audio or video, and whether it's MPEG-1 or MPEG-2:
        if ((stream_id&0xF0) == 0xE0) { // video
!       streamType = mpegVersion == 1 ? 1 : mpegVersion == 2 ? 2 : 0x10;
        } else if ((stream_id&0xE0) == 0xC0) { // audio
        streamType = mpegVersion == 1 ? 3 : mpegVersion == 2 ? 4 : 0xF;
        } else if (stream_id == 0xBD) { // private_stream1 (usually AC-3)
--- 109,115 ----
        // Instead, set the stream's type to default values, based on
whether
        // the stream is audio or video, and whether it's MPEG-1 or MPEG-2:
        if ((stream_id&0xF0) == 0xE0) { // video
!       streamType = mpegVersion == 1 ? 1 : mpegVersion == 2 ? 2 :
mpegVersion == 4 ? 0x10 : 0x1B;
        } else if ((stream_id&0xE0) == 0xC0) { // audio
        streamType = mpegVersion == 1 ? 3 : mpegVersion == 2 ? 4 : 0xF;
        } else if (stream_id == 0xBD) { // private_stream1 (usually AC-3)
***************
*** 121,127 ****

      if (fPCR_PID == 0) { // set it to this stream, if it's appropriate:
        if ((!fHaveVideoStreams && (streamType == 3 || streamType == 4 ||
streamType == 0xF))/* audio stream */ ||
!         (streamType == 1 || streamType == 2 || streamType == 0x10)/* video
stream */) {
        fPCR_PID = fCurrentPID; // use this stream's SCR for PCR
        }
      }
--- 121,127 ----

      if (fPCR_PID == 0) { // set it to this stream, if it's appropriate:
        if ((!fHaveVideoStreams && (streamType == 3 || streamType == 4 ||
streamType == 0xF))/* audio stream */ ||
!         (streamType == 1 || streamType == 2 || streamType == 0x10 ||
streamType == 0x1B)/* video stream */) {
        fPCR_PID = fCurrentPID; // use this stream's SCR for PCR
        }
      }
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From stas.oskin at gmail.com  Wed May 27 02:27:27 2009
From: stas.oskin at gmail.com (Stas Oskin)
Date: Wed, 27 May 2009 12:27:27 +0300
Subject: [Live-devel] openRTSP woes
In-Reply-To: <f06240803c642aa77f3db@66.80.62.44>
References: <A4B2EFAC-4B61-407B-BBA3-48CD00FFA22D@walking-productions.com>
	<f06240800c63a5c37d69c@66.80.62.44>
	<77938bc20905250157x241b9d00q937b51d9b677ded3@mail.gmail.com>
	<f06240803c642aa77f3db@66.80.62.44>
Message-ID: <77938bc20905270227x19e154f8j7756b05e5e6bc325@mail.gmail.com>

Hi.

What is the most recommended solution then in such case?

Keep video in elementary, but audio in mp3 for example, and sync between
them?

Regards.

2009/5/27 Ross Finlayson <finlayson at live555.com>

> 1) Just to verify, does it means that elementary stream files (m4e or m4v)
>> are unable to contain the PTS/DTS timestamps at all?
>>
>
> The video elementary stream files might contain these timestamps; the audio
> elementary files typically won't.
>
> --
>
> Ross Finlayson
> Live Networks, Inc.
> http://www.live555.com/
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
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From electrical_king at hotmail.com  Wed May 27 04:22:55 2009
From: electrical_king at hotmail.com (=?windows-1256?B?2uHHwSDayM8gx+Hkx9XRINrSyg==?=)
Date: Wed, 27 May 2009 14:22:55 +0300
Subject: [Live-devel] Meadia Server on VS 2005 or 2008
Message-ID: <COL106-W2841E39035D9F437FCA44695530@phx.gbl>


Hi,

I'm new to this mailing list but I didn't find an answer to my question in the last threads.

My question is:

I want to compile and debug the media server project using VS2005 or 2008 to generate the mediaserver.exe that run successfully with no caches.

Really I need to know how to make all projects in one solution.

I already tried these but also failed

http://metalkin.egloos.com/4478919

http://lists.live555.com/pipermail/live-devel/2008-January/008027.html

Please help ASAP.

Thanks in advance.

Alaa

 








_________________________________________________________________
Show them the way! Add maps and directions to your party invites. 
http://www.microsoft.com/windows/windowslive/products/events.aspx
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From gampa.harsha at gmail.com  Wed May 27 06:09:28 2009
From: gampa.harsha at gmail.com (harsha gampa)
Date: Wed, 27 May 2009 18:39:28 +0530
Subject: [Live-devel] RTP ports used by live555MediaServer
Message-ID: <27116a4e0905270609s7f090804y2382e95083030b8e@mail.gmail.com>

Hi ,

I would like to know the ports used by live555 server to stream the RTP
packets .i.e the port number range on the server which streams the RTP
packets.

Regards,
Harsha Gampa
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From finlayson at live555.com  Wed May 27 07:44:17 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 27 May 2009 07:44:17 -0700
Subject: [Live-devel] H264 TS
In-Reply-To: <92a42b330905270639mb3cb88dx13ef1473c2af1b90@mail.gmail.com>
References: <92a42b330905270639mb3cb88dx13ef1473c2af1b90@mail.gmail.com>
Message-ID: <f06240800c64301ee741f@[66.80.62.44]>

>Hi Ross,
>
>in attach my modified transport stream multiplexor to handle h264 
>stream type ...
>
>It works for me ... Is this correct ?

What value of "mpegVersion" to you use for H.264 video?  (A value of 
"4" should be used for regular MPEG-4 video, not H/.264.)

-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From finlayson at live555.com  Wed May 27 08:53:24 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 27 May 2009 08:53:24 -0700
Subject: [Live-devel] RTP ports used by live555MediaServer
In-Reply-To: <27116a4e0905270609s7f090804y2382e95083030b8e@mail.gmail.com>
References: <27116a4e0905270609s7f090804y2382e95083030b8e@mail.gmail.com>
Message-ID: <f06240801c64311430c11@[66.80.62.44]>

>I would like to know the ports used by live555 server to stream the 
>RTP packets .i.e the port number range on the server which streams 
>the RTP packets.

By default, RTP/RTCP ports start at 6970.  (Even numbers are used for 
RTP; odd numbers are used for RTCP.)

There's no 'range' of port numbers; the server uses however many port 
numbers it needs (but starting at 6970).
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From zmax.linkedin at gmail.com  Wed May 27 09:14:37 2009
From: zmax.linkedin at gmail.com (Massimo Zito)
Date: Wed, 27 May 2009 18:14:37 +0200
Subject: [Live-devel] H264 TS
In-Reply-To: <92a42b330905270911h5de1f41ic7b5f63859913833@mail.gmail.com>
References: <92a42b330905270639mb3cb88dx13ef1473c2af1b90@mail.gmail.com>
	<f06240800c64301ee741f@66.80.62.44>
	<92a42b330905270902q1c5e1831p9cb6c3ffda1d2cc5@mail.gmail.com>
	<92a42b330905270911h5de1f41ic7b5f63859913833@mail.gmail.com>
Message-ID: <92a42b330905270914o404758b0t6409552011539d69@mail.gmail.com>

2009/5/27 Massimo Zito <zmax.linkedin at gmail.com>

> sorry ... there's a mistake ... :)
>
> In original version:
>
> mpegVersion == 1 -> stream type -> 1
> mpegVersion == 2 -> stream type -> 2
> else stream type -> 0x10
>
> In my modified version:
>
> mpegVersion == 1 -> stream type -> 1
> mpegVersion == 2 -> stream type -> 2
> mpegVersion == 4 -> stream type -> 0x10
> else stream type -> 0x1b
>
> I use mpegVersion == 0x1b ...
>
>
>
>
> 2009/5/27 Massimo Zito <zmax.linkedin at gmail.com>
>
> In original version:
>>
>> mpegVersion == 1 -> stream type -> 1
>> mpegVersion == 2 -> stream type -> 2
>> else stream type -> 4
>>
>> In my modified version:
>>
>> mpegVersion == 1 -> stream type -> 1
>> mpegVersion == 2 -> stream type -> 2
>> mpegVersion == 4 -> stream type -> 0x10
>> else stream type -> 0x1b
>>
>> I use mpegVersion == 0x1b ...
>>
>>
>> 2009/5/27 Ross Finlayson <finlayson at live555.com>
>>
>>  Hi Ross,
>>>>
>>>> in attach my modified transport stream multiplexor to handle h264 stream
>>>> type ...
>>>>
>>>> It works for me ... Is this correct ?
>>>>
>>>
>>> What value of "mpegVersion" to you use for H.264 video?  (A value of "4"
>>> should be used for regular MPEG-4 video, not H/.264.)
>>>
>>> --
>>>
>>> Ross Finlayson
>>> Live Networks, Inc.
>>> http://www.live555.com/
>>> _______________________________________________
>>> live-devel mailing list
>>> live-devel at lists.live555.com
>>> http://lists.live555.com/mailman/listinfo/live-devel
>>>
>>
>>
>
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From kidjan at gmail.com  Wed May 27 11:21:02 2009
From: kidjan at gmail.com (Jeremy Noring)
Date: Wed, 27 May 2009 11:21:02 -0700
Subject: [Live-devel] Meadia Server on VS 2005 or 2008
In-Reply-To: <COL106-W2841E39035D9F437FCA44695530@phx.gbl>
References: <COL106-W2841E39035D9F437FCA44695530@phx.gbl>
Message-ID: <bda4ea420905271121p492d932fxf036b3a1cf48ee1c@mail.gmail.com>

I've wanted VS solutions as well (the current windows build environment
leaves much to be desired, but I guess that's the beauty of open source:
should you so desire...), but I think you'll have to create a VS solution
with four projects for each of the libraries in Live555, and then link any
applications that use Live555 stuff against those libraries.

If I have time, I may try to do this today, since I need this sort of
functionality for debugging symbols, debug/release builds, x64 support,
integration into our project, etc.  I'll let you know if I have any success.

2009/5/27 ???? ??? ?????? ??? <electrical_king at hotmail.com>

>  Hi,
>
> I'm new to this mailing list but I didn't find an answer to my question in
> the last threads.
>
> My question is:
>
> I want to compile and debug the media server project using VS2005 or 2008
> to generate the mediaserver.exe that run successfully with no caches.
>
> Really I need to know how to make all projects in one solution.
>
> I already tried these but also failed
>
> http://metalkin.egloos.com/4478919
>
> http://lists.live555.com/pipermail/live-devel/2008-January/008027.html
>
> Please help ASAP.
>
> Thanks in advance.
>
> Alaa
>
>
>
> <http://www.softlock.net/>
>
>
>
>
>
> ------------------------------
> See all the ways you can stay connected to friends and family<http://www.microsoft.com/windows/windowslive/default.aspx>
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
>


-- 
Where are we going?
And why am I in this hand-basket?
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From joeflin at 126.com  Wed May 27 15:30:16 2009
From: joeflin at 126.com (joeflin)
Date: Thu, 28 May 2009 06:30:16 +0800 (CST)
Subject: [Live-devel] H264 TS
In-Reply-To: <f06240800c64301ee741f@[66.80.62.44]>
References: <f06240800c64301ee741f@[66.80.62.44]>
	<92a42b330905270639mb3cb88dx13ef1473c2af1b90@mail.gmail.com>
Message-ID: <25953553.589471243463416431.JavaMail.coremail@bj126app52.126.com>

What is a good H264-TS client? 



?2009-05-27?22:44:17?"Ross?Finlayson"?<finlayson at live555.com>????
>>Hi?Ross,
>>
>>in?attach?my?modified?transport?stream?multiplexor?to?handle?h264?
>>stream?type?...
>>
>>It?works?for?me?...?Is?this?correct??
>
>What?value?of?"mpegVersion"?to?you?use?for?H.264?video???(A?value?of?
>"4"?should?be?used?for?regular?MPEG-4?video,?not?H/.264.)
>
>--?
>
>Ross?Finlayson
>Live?Networks,?Inc.
>http://www.live555.com/
>_______________________________________________
>live-devel?mailing?list
>live-devel at lists.live555.com
>http://lists.live555.com/mailman/listinfo/live-devel
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From finlayson at live555.com  Wed May 27 18:27:47 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 27 May 2009 18:27:47 -0700
Subject: [Live-devel] H264 TS
In-Reply-To: <25953553.589471243463416431.JavaMail.coremail@bj126app52.126.com>
References: <f06240800c64301ee741f@[66.80.62.44]>
	<92a42b330905270639mb3cb88dx13ef1473c2af1b90@mail.gmail.com>
	<25953553.589471243463416431.JavaMail.coremail@bj126app52.126.com>
Message-ID: <f06240801c6439902bca9@[66.80.62.44]>

>What is a good H264-TS client?

VLC should work.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From finlayson at live555.com  Wed May 27 18:36:22 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 27 May 2009 18:36:22 -0700
Subject: [Live-devel] H264 TS
In-Reply-To: <92a42b330905270639mb3cb88dx13ef1473c2af1b90@mail.gmail.com>
References: <92a42b330905270639mb3cb88dx13ef1473c2af1b90@mail.gmail.com>
Message-ID: <f06240802c6439aff33e8@[66.80.62.44]>

>in attach my modified transport stream multiplexor to handle h264 
>stream type ...

Thanks.  This will be included in the next release of the software.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From patbob at imoveinc.com  Thu May 28 09:46:46 2009
From: patbob at imoveinc.com (Patrick White)
Date: Thu, 28 May 2009 09:46:46 -0700
Subject: [Live-devel] Meadia Server on VS 2005 or 2008
In-Reply-To: <bda4ea420905271121p492d932fxf036b3a1cf48ee1c@mail.gmail.com>
References: <COL106-W2841E39035D9F437FCA44695530@phx.gbl>
	<bda4ea420905271121p492d932fxf036b3a1cf48ee1c@mail.gmail.com>
Message-ID: <200905280946.46730.patbob@imoveinc.com>

On Wednesday 27 May 2009 11:21 am, Jeremy Noring wrote:
> If I have time, I may try to do this today, since I need this sort of
> functionality for debugging symbols, debug/release builds, x64 support,
> integration into our project, etc.  I'll let you know if I have any
> success.

We needed the same thing, and found a way to generate the PDB files as live555 
builds.  Unfortunately, it required mods to the shared build files.

Basically, the changes were:

*) add something like "/Fd$(LIBNAME)D.pdb" to the COMPILE_OPTS in the 
win32config file.  We made two win32config files -- one that's used for 
release and doesn't have the option (because we don't want/need PDB files for 
release builds) and one for debug that does include it.

*) Modify Makefile.head in each project to set a LIBNAME variable to the name 
of the library being generated.  Here's an example from the groupsock 
project:
LIBNAME=libgroupsock

Recompile and you'll get PDB files generated next to the libraries.  We move 
them all into a libs directory for convenience.

*) To use them, I ended up copying them into the dir where we have the 
compiler put all the generated DLLs, libs and EXEs.


What I found was that the PDB files could not be renamed.  They could be 
moved, but must remain by their original name in oprder for Visual Studio to 
be able to use them.  That's why I had to modify the Makefile.head files to 
have a different LIBNAME in each -- I couldn't generate the PDB files by the 
same name for each project and later rename them, and there seemed to be no 
other place to quickly and easily get the lib name from.  The "D" on the end 
of the PDB file name is not really needed, its an artifact of our dev 
procedures.

hope that helps,
patbob

From auscaster at gmail.com  Thu May 28 05:46:42 2009
From: auscaster at gmail.com (Kam Low)
Date: Thu, 28 May 2009 22:46:42 +1000
Subject: [Live-devel] Meadia Server on VS 2005 or 2008
In-Reply-To: <bda4ea420905271121p492d932fxf036b3a1cf48ee1c@mail.gmail.com>
References: <COL106-W2841E39035D9F437FCA44695530@phx.gbl>
	<bda4ea420905271121p492d932fxf036b3a1cf48ee1c@mail.gmail.com>
Message-ID: <e23b37dc0905280546y3973c5d3w3197fb9ece4e69c9@mail.gmail.com>

I spent a day of and hair pulling, keyboard bashing and shouting of
obscenities trying to do this very thing about a week ago.

Hopefully the fruits of my labor will save someone from the same ordeal :P

Please find the vs2008 project and compiled libraries attached.

Regards,
Kam



2009/5/28 Jeremy Noring <kidjan at gmail.com>

> I've wanted VS solutions as well (the current windows build environment
> leaves much to be desired, but I guess that's the beauty of open source:
> should you so desire...), but I think you'll have to create a VS solution
> with four projects for each of the libraries in Live555, and then link any
> applications that use Live555 stuff against those libraries.
>
> If I have time, I may try to do this today, since I need this sort of
> functionality for debugging symbols, debug/release builds, x64 support,
> integration into our project, etc.  I'll let you know if I have any success.
>
> 2009/5/27 ???? ??? ?????? ??? <electrical_king at hotmail.com>
>
>>  Hi,
>>
>> I'm new to this mailing list but I didn't find an answer to my question in
>> the last threads.
>>
>> My question is:
>>
>> I want to compile and debug the media server project using VS2005 or 2008
>> to generate the mediaserver.exe that run successfully with no caches.
>>
>> Really I need to know how to make all projects in one solution.
>>
>> I already tried these but also failed
>>
>> http://metalkin.egloos.com/4478919
>>
>> http://lists.live555.com/pipermail/live-devel/2008-January/008027.html
>>
>> Please help ASAP.
>>
>> Thanks in advance.
>>
>> Alaa
>>
>>
>>
>> <http://www.softlock.net/>
>>
>>
>>
>>
>>
>> ------------------------------
>> See all the ways you can stay connected to friends and family<http://www.microsoft.com/windows/windowslive/default.aspx>
>>
>> _______________________________________________
>> live-devel mailing list
>> live-devel at lists.live555.com
>> http://lists.live555.com/mailman/listinfo/live-devel
>>
>>
>
>
> --
> Where are we going?
> And why am I in this hand-basket?
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
>
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From electrical_king at hotmail.com  Thu May 28 14:10:38 2009
From: electrical_king at hotmail.com (=?windows-1256?B?2uHHwSDayM8gx+Hkx9XRINrSyg==?=)
Date: Fri, 29 May 2009 00:10:38 +0300
Subject: [Live-devel] Meadia Server on VS 2005 or 2008
In-Reply-To: <200905280946.46730.patbob@imoveinc.com>
References: <COL106-W2841E39035D9F437FCA44695530@phx.gbl>
	<bda4ea420905271121p492d932fxf036b3a1cf48ee1c@mail.gmail.com> 
	<200905280946.46730.patbob@imoveinc.com>
Message-ID: <COL106-W21DB739801B9ABBC4BB73995500@phx.gbl>


Hi,

thanks much for your help

I used the instructions here 

http://www.cs.tut.fi/~hirvone2/SGN_5106_Exercise4_2009.pdf

to compile and debug the media server and it run successfully But in the test with any rtsp link it caches !!

Does anyone know the reason ?

Thanks.









> From: patbob at imoveinc.com
> To: live-devel at ns.live555.com
> Date: Thu, 28 May 2009 09:46:46 -0700
> Subject: Re: [Live-devel] Meadia Server on VS 2005 or 2008
> 
> On Wednesday 27 May 2009 11:21 am, Jeremy Noring wrote:
> > If I have time, I may try to do this today, since I need this sort of
> > functionality for debugging symbols, debug/release builds, x64 support,
> > integration into our project, etc.  I'll let you know if I have any
> > success.
> 
> We needed the same thing, and found a way to generate the PDB files as live555 
> builds.  Unfortunately, it required mods to the shared build files.
> 
> Basically, the changes were:
> 
> *) add something like "/Fd$(LIBNAME)D.pdb" to the COMPILE_OPTS in the 
> win32config file.  We made two win32config files -- one that's used for 
> release and doesn't have the option (because we don't want/need PDB files for 
> release builds) and one for debug that does include it.
> 
> *) Modify Makefile.head in each project to set a LIBNAME variable to the name 
> of the library being generated.  Here's an example from the groupsock 
> project:
> LIBNAME=libgroupsock
> 
> Recompile and you'll get PDB files generated next to the libraries.  We move 
> them all into a libs directory for convenience.
> 
> *) To use them, I ended up copying them into the dir where we have the 
> compiler put all the generated DLLs, libs and EXEs.
> 
> 
> What I found was that the PDB files could not be renamed.  They could be 
> moved, but must remain by their original name in oprder for Visual Studio to 
> be able to use them.  That's why I had to modify the Makefile.head files to 
> have a different LIBNAME in each -- I couldn't generate the PDB files by the 
> same name for each project and later rename them, and there seemed to be no 
> other place to quickly and easily get the lib name from.  The "D" on the end 
> of the PDB file name is not really needed, its an artifact of our dev 
> procedures.
> 
> hope that helps,
> patbob
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel

_________________________________________________________________
More than messages?check out the rest of the Windows Live?.
http://www.microsoft.com/windows/windowslive/
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From finlayson at live555.com  Thu May 28 15:03:43 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Thu, 28 May 2009 15:03:43 -0700
Subject: [Live-devel] Meadia Server on VS 2005 or 2008
In-Reply-To: <e23b37dc0905280546y3973c5d3w3197fb9ece4e69c9@mail.gmail.com>
References: <COL106-W2841E39035D9F437FCA44695530@phx.gbl>
	<bda4ea420905271121p492d932fxf036b3a1cf48ee1c@mail.gmail.com>
	<e23b37dc0905280546y3973c5d3w3197fb9ece4e69c9@mail.gmail.com>
Message-ID: <f06240800c644b97ba529@[66.80.62.44]>

>I spent a day of and hair pulling, keyboard bashing and shouting of 
>obscenities trying to do this very thing about a week ago.
>
>Hopefully the fruits of my labor will save someone from the same ordeal :P
>
>Please find the vs2008 project and compiled libraries attached.


As I've noted in the past, contributions like this are welcome, but 
unfortunately we can't really also make them available on our own web 
site, because there's no way to update them whenever our source files 
change (e.g., by adding, removing, or renaming source files).

The only thing that we could support would be some sort of script 
that could generate 'project files' automaticallly from the ".mak" or 
"Makefile.tail" files.

Makefiles have been the standard software configuration mechanism for 
almost 30 years now.  If recent versions of Microsoft's development 
tools no longer support Makefiles, then I consider that to be 
Microsoft's problem, not ours.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

From auscaster at gmail.com  Thu May 28 18:26:47 2009
From: auscaster at gmail.com (Kam Low)
Date: Fri, 29 May 2009 11:26:47 +1000
Subject: [Live-devel] Meadia Server on VS 2005 or 2008
In-Reply-To: <f06240800c644b97ba529@66.80.62.44>
References: <COL106-W2841E39035D9F437FCA44695530@phx.gbl>
	<bda4ea420905271121p492d932fxf036b3a1cf48ee1c@mail.gmail.com>
	<e23b37dc0905280546y3973c5d3w3197fb9ece4e69c9@mail.gmail.com>
	<f06240800c644b97ba529@66.80.62.44>
Message-ID: <e23b37dc0905281826m14c60e4dg7b9757877e4a2f9d@mail.gmail.com>

I couldn't agree more - MS continues to create problems for developers
rather than solve them.

I believe vs2008 still supports Makefiles though, but I was unsuccessful
compiling the latest libraries using the genWindowsMakefiles.cmd method.
Cygwin cross-compiling and Mingw both had issues to, somewhere around
rtcp_from_spec I think.

This was probably due to my lack of understanding though.

Anyway thanks for the great software, you have done a fantastic job!

Regards,
Kam


2009/5/29 Ross Finlayson <finlayson at live555.com>

> I spent a day of and hair pulling, keyboard bashing and shouting of
>> obscenities trying to do this very thing about a week ago.
>>
>> Hopefully the fruits of my labor will save someone from the same ordeal :P
>>
>> Please find the vs2008 project and compiled libraries attached.
>>
>
>
> As I've noted in the past, contributions like this are welcome, but
> unfortunately we can't really also make them available on our own web site,
> because there's no way to update them whenever our source files change
> (e.g., by adding, removing, or renaming source files).
>
> The only thing that we could support would be some sort of script that
> could generate 'project files' automaticallly from the ".mak" or
> "Makefile.tail" files.
>
> Makefiles have been the standard software configuration mechanism for
> almost 30 years now.  If recent versions of Microsoft's development tools no
> longer support Makefiles, then I consider that to be Microsoft's problem,
> not ours.
> --
>
> Ross Finlayson
> Live Networks, Inc.
> http://www.live555.com/
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
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From hoanglinh9466 at gmail.com  Thu May 28 21:40:32 2009
From: hoanglinh9466 at gmail.com (hoang linh)
Date: Fri, 29 May 2009 11:40:32 +0700
Subject: [Live-devel] Found document about RTP payload format of ASF
Message-ID: <742313b60905282140y63f9be34o3043d1c661b9fc33@mail.gmail.com>

Hi !
I know that live555 doesn't support X-ASF-PF format. And I 've found a
document (http://go.microsoft.com/fwlink/?LinkId=89814) here (
http://msdn.microsoft.com/en-us/library/cc245257(PROT.10).aspx). It mention
about ASF file and also RTP payload format of ASF. I think it will be useful
for anyone want to write code to handle this.

And one more thing, live555 doesn't support ASF, so why VLC player use
live555 library and it can play ASF ???
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From finlayson at live555.com  Fri May 29 01:35:18 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Fri, 29 May 2009 01:35:18 -0700
Subject: [Live-devel] Found document about RTP payload format of ASF
In-Reply-To: <742313b60905282140y63f9be34o3043d1c661b9fc33@mail.gmail.com>
References: <742313b60905282140y63f9be34o3043d1c661b9fc33@mail.gmail.com>
Message-ID: <f06240800c6454df7bc14@[66.80.62.44]>

>I know that live555 doesn't support X-ASF-PF format. And I 've found 
>a document 
>(<http://go.microsoft.com/fwlink/?LinkId=89814>http://go.microsoft.com/fwlink/?LinkId=89814) 
>here 
>(<http://msdn.microsoft.com/en-us/library/cc245257(PROT.10).aspx>http://msdn.microsoft.com/en-us/library/cc245257(PROT.10).aspx). 
>It mention about ASF file and also RTP payload format of ASF. I 
>think it will be useful for anyone want to write code to handle this.

Yes, if anyone were to implement this RTP payload format, I would 
likely add it to the released code.

>
>And one more thing, live555 doesn't support ASF, so why VLC player 
>use live555 library and it can play ASF ???

Most likely because - in this case - VLC is downloading the file 
using HTTP, rather than RTSP.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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From bourdag at gmail.com  Fri May 29 03:03:08 2009
From: bourdag at gmail.com (bourda guillaume)
Date: Fri, 29 May 2009 12:03:08 +0200
Subject: [Live-devel] Using SimpleRTPSource and SimpleRTPSink to record and
	re-stream H264
Message-ID: <c0a55910905290303k5ac1cc50kfe1d3be14e2d62fd@mail.gmail.com>

Hi Ross,

I want want to record various [audio+video] streams  (m4v, m4a, h264,
pcmu...) on my computer and then re-stream it later.
I would like to set a code that works for all types of video and audio
formats. Basically I would receive RTP packets, store their content without
looking at it, and re-send the RTP packets later.

I figured out that the best way to do that was to use SimpleRTPSource and
SimpleRTPSink while storing all SDP and presentation times related infos.
It seems to work fine for a M4V/M4A(MPEG4-GENERIC) stream but doesn't work
for an H264 stream: when I re-stream H264, the client says "unsupported NAL
type for H264".

Can you give me some hint on what I should look for?
Thanks in advance
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From tech.siya at gmail.com  Sat May 30 22:33:41 2009
From: tech.siya at gmail.com (siyara nt)
Date: Sun, 31 May 2009 11:03:41 +0530
Subject: [Live-devel] Building live555 for WinCE 6.0 based DM355 target
Message-ID: <ca5766a70905302233k323f5d2bu590d2cd338faad2a@mail.gmail.com>

Hi,

I am newbie to this forum. I am running WinCE 6.0 on DM355.
I am trying to build Live555 on WinCE 6.0 for DM355(ARM) target.
If anyone has already tried this, please help.

I'm using platform builder toolkit with WinCE and VS2005 IDE.

thanks
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From xujin at fun.21cn.com  Sun May 31 00:09:25 2009
From: xujin at fun.21cn.com (xujin)
Date: Sun, 31 May 2009 15:09:25 +0800
Subject: [Live-devel] about MPEG1or2Demux
Message-ID: <61C3D3DF7BC94D45B1A6DDC13057FC4D@covondxujin>

Hi,  
  I test testMPEG1or2Splitter.cpp, but it doesn't work well. 
I make a program stream file by vlc, as to be the input file, the sizeof
output video file and and video file are zero.
I don't  know why. Anything wrong?
 
   Thanks.
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From finlayson at live555.com  Sun May 31 00:56:43 2009
From: finlayson at live555.com (Ross Finlayson)
Date: Sun, 31 May 2009 00:56:43 -0700
Subject: [Live-devel] about MPEG1or2Demux
In-Reply-To: <61C3D3DF7BC94D45B1A6DDC13057FC4D@covondxujin>
References: <61C3D3DF7BC94D45B1A6DDC13057FC4D@covondxujin>
Message-ID: <f06240802c647e87662b3@[66.80.62.44]>

>Hi, 
>   I test testMPEG1or2Splitter.cpp, but it doesn't work well.
>I make a program stream file by vlc, as to be the input file, the 
>sizeof output video file and and video file are zero.
>I don't  know why. Anything wrong?

Please put your Program Stream file on a publically-accessible web 
(or FTP) server, and post the URL (not the file itself) to this 
mailing list, and we'll take a look at it, to see if we can figure 
out what's wrong.
-- 

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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From electrical_king at hotmail.com  Sun May 31 10:41:45 2009
From: electrical_king at hotmail.com (=?windows-1256?B?2uHHwSDayM8gx+Hkx9XRINrSyg==?=)
Date: Sun, 31 May 2009 20:41:45 +0300
Subject: [Live-devel] Include external libraries ?
Message-ID: <COL106-W109C8BDD8E7C0AD92DDE9C954F0@phx.gbl>


Hi,

I'm now using the live media server on Windows XP and VS2005 and I want to include some external libraries to use it in my code from that path 

C:\Program Files\Microsoft Visual Studio 8\VC\PlatformSDK\Include

like sql.h sqlext.h

how can I make that?

thanks in advance









_________________________________________________________________
Drag n? drop?Get easy photo sharing with Windows Live? Photos.

http://www.microsoft.com/windows/windowslive/products/photos.aspx
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