[Live-devel] live-devel Digest, Vol 45, Issue 11

cheung bonheur bonheur.cheung at seldes.com
Fri Jul 13 00:38:48 PDT 2007


Hello,

I want to record a stream rtsp to file using openrtsp
But when i read this file, VLC cant read it...

Thanks foryour help

----- Original Message ----- 
From: <live-devel-request at ns.live555.com>
To: <live-devel at ns.live555.com>
Sent: Friday, July 13, 2007 6:33 AM
Subject: live-devel Digest, Vol 45, Issue 11


> Send live-devel mailing list submissions to
> live-devel at lists.live555.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.live555.com/mailman/listinfo/live-devel
> or, via email, send a message with subject or body 'help' to
> live-devel-request at lists.live555.com
>
> You can reach the person managing the list at
> live-devel-owner at lists.live555.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of live-devel digest..."
>
>
> Today's Topics:
>
>   1. Re: SDP file of a multiple destination session (Ross Finlayson)
>   2. Re: SDP file of a multiple destination session
>      (Ramon Martin de Pozuelo Genis)
>   3. Re: SDP file of a multiple destination session
>      (Ramon Martin de Pozuelo Genis)
>   4. Re: SDP file of a multiple destination session (Ross Finlayson)
>   5. RTP Sequence number - alternating values (Christian Frahm)
>   6. Re: SDP file of a multiple destination session
>      (Ramon Martin de Pozuelo Genis)
>   7. Re: RTP Sequence number - alternating values (Ross Finlayson)
>   8. Re: RTCP and synchronization (Ross Finlayson)
>   9. Live555 Media Server (Jimmy A Samaha)
>  10. nitin.e wants to talk to you on Yoomba (nitin.e)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Thu, 12 Jul 2007 03:37:54 -0700
> From: Ross Finlayson <finlayson at live555.com>
> Subject: Re: [Live-devel] SDP file of a multiple destination session
> To: LIVE555 Streaming Media - development & use
> <live-devel at ns.live555.com>
> Message-ID: <f06240807c2bbb3dd2917@[66.80.62.44]>
> Content-Type: text/plain; charset="us-ascii" ; format="flowed"
>
> You probably shouldn't be using "addDestination()" - that is a
> specialized function used only to implement on-demend unicast
> streaming to multiple clients from a single source.  (Note that, for
> unicast on-demand streams, the SDP description should contain the
> special address 0.0.0.0, not a specific unicast address.)
>
> I don't really know what you are trying to do, but I suspect that you
> should just leave the existing code as it is, and use it without
> modification.
> -- 
>
> Ross Finlayson
> Live Networks, Inc.
> http://www.live555.com/
>
>
> ------------------------------
>
> Message: 2
> Date: Thu, 12 Jul 2007 13:00:44 +0200 (CEST)
> From: "Ramon Martin de Pozuelo Genis" <tl11305 at salle.url.edu>
> Subject: Re: [Live-devel] SDP file of a multiple destination session
> To: "LIVE555 Streaming Media - development & use"
> <live-devel at ns.live555.com>
> Message-ID:
> <1995.172.16.11.128.1184238044.squirrel at webmail.salle.url.edu>
> Content-Type: text/plain;charset=iso-8859-1
>
>> You probably shouldn't be using "addDestination()" - that is a
>> specialized function used only to implement on-demend unicast
>> streaming to multiple clients from a single source.  (Note that, for
>> unicast on-demand streams, the SDP description should contain the
>> special address 0.0.0.0, not a specific unicast address.)
>
> It's exactly my case, I create a service that streams on broadcast one
> description of H.264 SVC and the second description is streaming on-demand
> unicast, but I didn't know that SDP should contain 0.0.0.0 to this 
> address,
> thank you very much Ross.
>
>> I don't really know what you are trying to do, but I suspect that you
>> should just leave the existing code as it is, and use it without
>> modification.
>> --
>
>
> Well, I added some classes like my own class dervated from
> H264RTPVideoStreamer and some classes that make similar functions to
> testMPEG4VideoStreamer, but using my own H.264 source and using UDP 
> commands
> from a web service to add and remove sessions from RTSPServer. I want to 
> use 3
> types of services: broadcast H.264 service (all the descriptions), on 
> demand
> H.264 AVC service, the last one that I commented above. All the existing 
> code
> is not modificated, only my added classes. But if you are interested in it 
> I
> could share with you my project. There are some universities of europe
> involved and I'm doing the playout part (videoserver) and some partners of 
> my
> department the web service to control it.
>
> Thanks again Ross for all your attention,
>
> Ramon
>
>
>
> ------------------------------
>
> Message: 3
> Date: Thu, 12 Jul 2007 13:12:54 +0200 (CEST)
> From: "Ramon Martin de Pozuelo Genis" <tl11305 at salle.url.edu>
> Subject: Re: [Live-devel] SDP file of a multiple destination session
> To: "LIVE555 Streaming Media - development & use"
> <live-devel at ns.live555.com>
> Message-ID:
> <2007.172.16.11.128.1184238774.squirrel at webmail.salle.url.edu>
> Content-Type: text/plain;charset=iso-8859-1
>
> Sorry, I forgot ask you about SDP file. Now address is correct (0.0.0.0) 
> but
> what about port? "m" line of SDP is still using port I specified when I 
> create
> groupsock, could I change it? Should I change it? If the first description 
> is
> sending broadcast in te same port it will cause a problem.
>
> Thanks again,
>
> Ramon
>
>
>
>
> ------------------------------
>
> Message: 4
> Date: Thu, 12 Jul 2007 04:54:49 -0700
> From: Ross Finlayson <finlayson at live555.com>
> Subject: Re: [Live-devel] SDP file of a multiple destination session
> To: LIVE555 Streaming Media - development & use
> <live-devel at ns.live555.com>
> Message-ID: <f06240809c2bbc60930d5@[66.80.62.44]>
> Content-Type: text/plain; charset="us-ascii" ; format="flowed"
>
> Again, it sounds like you're trying to reinvent the wheel.  The
> "OnDemandServerMediaSubsession" class works just fine - you should
> just use it (by defining your own subclass).  Note the several
> examples in the code.  You should be using "testOnDemandRTSPServer" -
> not "testMPEG4VideoStreamer" - as a model for your application.
>
> -- 
>
> Ross Finlayson
> Live Networks, Inc.
> http://www.live555.com/
>
>
> ------------------------------
>
> Message: 5
> Date: Thu, 12 Jul 2007 15:46:56 +0200
> From: "Christian Frahm" <frahm.c at googlemail.com>
> Subject: [Live-devel] RTP Sequence number - alternating values
> To: live-devel at ns.live555.com
> Message-ID:
> <8039fa140707120646k49f0f961w3eeb211ed2d74014 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Thanks for the answer Ross. I will make myself a bit more clear to state 
> the
> case.
>
> I am streaming a sequence of PES (both audio and video). I use the demux
> class to stream audio and video separatelly ( audio to IP port 7777 and
> video to 8888). I am using a dedicated network - so all traffic to that 
> port
> must  be generated by the LiveMedia server.
>
> At port 7777 - audio can be received and decoded successfully.
>
> At port 8888, video was "jiggy". So ran some network traces with 
> Wireshark.
> RTP packets DO get sent in port 8888. However, as I have stated before, 
> the
> RTP sequence number is not continuous - and alternates. Secondly - I have
> also observed some RTP packets in port 8888 with payload 72!
>
> With video - the Demux is connected to a discrete framer, which is 
> connected
> to a RTP Sink (just like the VOB Streamer code...).
>
> Does anyone have any idea why this could happen? Why is my RTP sink
> generating bad sequence numbers? What about the RTP packets with payload 
> 72!
> They must be coming from the same RTPSink and Framer... but how can that
> happen!?
>
> Thanks!
>
>
>
> -------- Original Message -------------------
>>Audio works fine - I am encountering a rather peculiar error with
>>video. My video RTP packets are being send with alternating Sequence
>>Numbers. For example:
>>
>>RTP packet 1 Sequence number:54820
>>RTP packet 2 Sequence number:44758
>>RTP packet 3 Sequence number:54821
>>RTP packet 4 Sequence number:44759
>>RTP packet 5 Sequence number:54822
>>RTP packet 6 Sequence number:44760
>>RTP packet 7 Sequence number:54823
>>
>>It is as if it sees two video streams when in fact there should only be
> one.
>>
>>Can anyone tell me how the Sequence Number is calculated?
>
> Sequentially for each (video or audio) RTP stream.
>
> Therefore, you must be generating more than one RTP stream.
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: 
> http://lists.live555.com/pipermail/live-devel/attachments/20070712/7f31e991/attachment-0001.html
>
> ------------------------------
>
> Message: 6
> Date: Thu, 12 Jul 2007 16:37:59 +0200 (CEST)
> From: "Ramon Martin de Pozuelo Genis" <tl11305 at salle.url.edu>
> Subject: Re: [Live-devel] SDP file of a multiple destination session
> To: "LIVE555 Streaming Media - development & use"
> <live-devel at ns.live555.com>
> Message-ID:
> <1066.172.16.11.128.1184251079.squirrel at webmail.salle.url.edu>
> Content-Type: text/plain;charset=iso-8859-1
>
>> Again, it sounds like you're trying to reinvent the wheel.  The
>> "OnDemandServerMediaSubsession" class works just fine - you should
>> just use it (by defining your own subclass).  Note the several
>> examples in the code.  You should be using "testOnDemandRTSPServer" -
>> not "testMPEG4VideoStreamer" - as a model for your application.
>
> I use "testOnDemandRTSPServer" as a model for the video on demand service, 
> but
> in this case I am sending broadcast all time and resending the 
> synchronized
> source (but description 2) to some unicast address (some kind of Quality 
> On
> Demand). Maybe I can use on demand model but I need to start reading
> description 2 source at same time I start reading description 1 and 
> sending it
> on broadcast, so I thought it was the easiest way (maybe not the most
> correct). Now it works fine, and I only have the problem in "m" line 
> creating
> automatically SDP file, but thanks for your observation, I will revise it 
> for
> a better implementation.
>
> Thanks anyway,
>
> Ramon
>
>
>
> ------------------------------
>
> Message: 7
> Date: Thu, 12 Jul 2007 09:11:59 -0700
> From: Ross Finlayson <finlayson at live555.com>
> Subject: Re: [Live-devel] RTP Sequence number - alternating values
> To: LIVE555 Streaming Media - development & use
> <live-devel at ns.live555.com>
> Message-ID: <f0624080ac2bc01fa42dc@[66.80.62.44]>
> Content-Type: text/plain; charset="us-ascii" ; format="flowed"
>
>>Does anyone have any idea why this could happen? Why is my RTP sink
>>generating bad sequence numbers? What about the RTP packets with
>>payload 72! They must be coming from the same RTPSink
>
> No.  Unless you have modified the existing library code, your
> "alternating sequence numbers" must be coming from two different
> "RTPSink" objects.
>
> You're going to have to figure out for yourself why your application
> code has created two or more "RTPSink" objects that (apparently) use
> the same "groupsock" object.  But Remember, You Have Complete Source
> Code.
> -- 
>
> Ross Finlayson
> Live Networks, Inc.
> http://www.live555.com/
>
>
> ------------------------------
>
> Message: 8
> Date: Thu, 12 Jul 2007 09:24:54 -0700
> From: Ross Finlayson <finlayson at live555.com>
> Subject: Re: [Live-devel] RTCP and synchronization
> To: LIVE555 Streaming Media - development & use
> <live-devel at ns.live555.com>
> Message-ID: <f0624080bc2bc039fa58f@[66.80.62.44]>
> Content-Type: text/plain; charset="us-ascii" ; format="flowed"
>
>>Ok, but I still don't get how the final recipient gets to know the
>>*original* presentation time (which correspond to audio) if I generate a
>>new one in the transcoder...
>
> Each incoming frame (from your "RTPSource" object) has an accurate
> presentation time, which you know (because it's passed as a parameter
> to your "afterGetting...()" function).  If your transcoder works by
> translating each incoming frame into a corresponding new transcoded
> frame, then it should simply assign each transcoded frame the
> presentation time from its corresponding incoming frame.  (Note,
> though, that if the incoming frames contain B-frames, but the
> transcoded frames don't, then the sequence of transcoded frames will
> not exactly match the sequence of incoming frames, and you should
> reorder the presentation times accordingly.)
>
> If, however, your transcoder does *not* work by translating each
> incoming frame into a corresponding new transcoded frame, then things
> are more complicated.
>
> But in any case, you're on your own from now on...
> -- 
>
> Ross Finlayson
> Live Networks, Inc.
> http://www.live555.com/
>
>
> ------------------------------
>
> Message: 9
> Date: Thu, 12 Jul 2007 18:44:21 -0400
> From: Jimmy A Samaha <jsamaha at MIT.EDU>
> Subject: [Live-devel] Live555 Media Server
> To: live-devel at ns.live555.com
> Message-ID: <20070712184421.ugnpq8exikkk8kss at webmail.mit.edu>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Hey all,
>
> I am trying to send live audio from a microphone through Media Server to 
> my
> client. Any ideas?
>
> Thanks
>
> Jimmy
>
>
> ------------------------------
>
> Message: 10
> Date: 12 Jul 2007 23:32:46 -0500
> From: "nitin.e" <nitin.e at gmail.com>
> Subject: [Live-devel] nitin.e wants to talk to you on Yoomba
> To: live-devel at ns.live555.com
> Message-ID: <18798979.1184301166394.JavaMail.SYSTEM at localhost>
> Content-Type: text/plain; charset="us-ascii"
>
> An HTML attachment was scrubbed...
> URL: 
> http://lists.live555.com/pipermail/live-devel/attachments/20070712/4a67109a/attachment.html
>
> ------------------------------
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
>
> End of live-devel Digest, Vol 45, Issue 11
> ******************************************
> 


More information about the live-devel mailing list