[Live-devel] playSIP - creates empty file
Markus Schumann
markuss at sonicfoundry.com
Thu May 2 11:52:02 PDT 2013
I am trying to record the audio from a Polycom Telepresence m100 SIP software client.
On 10.0.71.24 I run a software VTC client configure to use SIP (Polycom Telepresence m100).
The URI is sip:10.0.71.24 at 10.0.71.24
On 10.0.71.109 I run the live555 command line tool playSIP calling 10.0.71.24.
playSIP creates only a zero length file named: audio-PCMU-1
Any advice?
Thanks
Markus.
The connecting is established - output from playSIP:
==== START: playSIP debug output ==========================================================================
Sending request: INVITE sip:10.0.71.24 at 10.0.71.24 SIP/2.0
From: 10.0.71.24 <sip:10.0.71.24 at 10.0.71.109>;tag=4201176568
Via: SIP/2.0/UDP 10.0.71.109:64250
Max-Forwards: 70
To: sip:10.0.71.24 at 10.0.71.24
Contact: sip:10.0.71.24 at 10.0.71.109:64250
Call-ID: 2759522855 at 10.0.71.109
CSeq: 1 INVITE
Content-Type: application/sdp
User-Agent: Test_playSIP.exe (LIVE555 Streaming Media v2013.04.16)
Content-Length: 123
v=0
o=- 2759522855 0 IN IP4 10.0.71.109
s=Test_playSIP.exe session
c=IN IP4 10.0.71.109
t=0 0
m=audio 8000 RTP/AVP 0
Received INVITE response: SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.71.109:64250
From: "10.0.71.24" <sip:10.0.71.24 at 10.0.71.109>;tag=4201176568
To: "markuss" <sip:10.0.71.24 at 10.0.71.24>;tag=FA32ABC5-B2C76DF4
CSeq: 1 INVITE
Call-ID: 2759522855 at 10.0.71.109
Contact: <sip:markuss at 10.0.71.24:5060>
User-Agent: Polycom Telepresence m100/1.0.5.29417_4151
Content-Length: 0
Received INVITE response: SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.71.109:64250
From: "10.0.71.24" <sip:10.0.71.24 at 10.0.71.109>;tag=4201176568
To: "markuss" <sip:10.0.71.24 at 10.0.71.24>;tag=FA32ABC5-B2C76DF4
CSeq: 1 INVITE
Call-ID: 2759522855 at 10.0.71.109
Contact: <sip:markuss at 10.0.71.24:5060>
User-Agent: Polycom Telepresence m100/1.0.5.29417_4151
Content-Length: 0
Received INVITE response: SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.71.109:64250
From: "10.0.71.24" <sip:10.0.71.24 at 10.0.71.109>;tag=4201176568
To: "markuss" <sip:10.0.71.24 at 10.0.71.24>;tag=FA32ABC5-B2C76DF4
CSeq: 1 INVITE
Call-ID: 2759522855 at 10.0.71.109
Contact: <sip:markuss at 10.0.71.24:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRAC
K, UPDATE, REFER
User-Agent: Polycom Telepresence m100/1.0.5.29417_4151
Content-Type: application/sdp
Content-Length: 879
v=0
o=- 1367513200 1367513200 IN IP4 10.0.71.24
s=Polycom IP Phone
c=IN IP4 10.0.71.24
b=AS:64
t=0 0
m=audio 8000 RTP/AVP 118 115 114 113 99 98 97 102 101 103 9 15 0 8 119
a=sendrecv
a=rtpmap:118 SIRENLPR/16000
a=fmtp:118 bitrate=48000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:114 G7221/32000
a=fmtp:114 bitrate=32000
a=rtpmap:113 G7221/32000
a=fmtp:113 bitrate=24000
a=rtpmap:99 SIREN14/16000
a=fmtp:99 bitrate=48000
a=rtpmap:98 SIREN14/16000
a=fmtp:98 bitrate=32000
a=rtpmap:97 SIREN14/16000
a=fmtp:97 bitrate=24000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:101 G7221/16000
a=fmtp:101 bitrate=24000
a=rtpmap:103 G7221/16000
a=fmtp:103 bitrate=16000
a=rtpmap:9 G722/8000
a=fmtp:9 bitrate=64000
a=rtpmap:15 G728/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:119 telephone-event/8000
a=fmtp:119 0-15
Opened URL "sip:10.0.71.24 at 10.0.71.24", returning a SDP description:
v=0
o=- 2759522855 0 IN IP4 10.0.71.109
s=Test_playSIP.exe session
c=IN IP4 10.0.71.109
t=0 0
m=audio 8000 RTP/AVP 0
Created receiver for "audio/PCMU" subsession (client ports 8000-8001)
Setup "audio/PCMU" subsession (client ports 8000-8001)
Created output file: "audio-PCMU-1"
Sending request: ACK sip:10.0.71.24 at 10.0.71.24 SIP/2.0
From: 10.0.71.24 <sip:10.0.71.24 at 10.0.71.109>;tag=4201176568
Via: SIP/2.0/UDP 10.0.71.109:64250
Max-Forwards: 70
To: sip:10.0.71.24 at 10.0.71.24;tag=FA32ABC5-B2C76DF4
Call-ID: 2759522855 at 10.0.71.109
CSeq: 1 ACK
Content-Length: 0
Started playing session
Receiving streamed data (for up to 60.000000 seconds)...
==== END: playSIP debug output ==========================================================================
==== START: 10.0.71.109 (playSIP host) network log ============================================================
Captured on 10.0.71.109
17 1:34:56 PM 5/2/2013 1.4581485 10.0.71.109 10.0.71.24 SDP SDP:Request: INVITE sip:10.0.71.24 at 10.0.71.24 SIP/2.0; SDP:SessionName=Test_playSIP.exe session, Version=0, MediaDescription=audio 8000 RTP/AVP 0 {SIP:17, UDP:16, IPv4:15}
18 1:34:56 PM 5/2/2013 1.4617849 10.0.71.24 10.0.71.109 SIP SIP:Response: SIP/2.0 100 Trying {SIP:17, UDP:16, IPv4:15}
19 1:34:56 PM 5/2/2013 1.4716476 10.0.71.24 10.0.71.109 SIP SIP:Response: SIP/2.0 180 Ringing {SIP:17, UDP:16, IPv4:15}
42 1:34:59 PM 5/2/2013 4.6255978 10.0.71.24 10.0.71.109 SDP SDP:Response: SIP/2.0 200 OK; SDP:SessionName=Polycom IP Phone, Version=0, MediaDescription=audio 8000 RTP/AVP 118 115 114 113 99 98 97 102 101 103 9 15 0 8 119 {SIP:17, UDP:16, IPv4:15}
44 1:34:59 PM 5/2/2013 4.6385918 10.0.71.109 10.0.71.24 SIP SIP:Request: ACK sip:10.0.71.24 at 10.0.71.24 SIP/2.0 {SIP:17, UDP:16, IPv4:15}
50 1:35:00 PM 5/2/2013 5.5851924 10.0.71.24 10.0.71.109 RTP RTP:PayloadType = PCMU Audio, 8000Hz [1 Channel], SSRC = 242919752, Seq = 0, TimeStamp = 0 {UDP:38, IPv4:15}
52 1:35:00 PM 5/2/2013 5.6130255 10.0.71.24 10.0.71.109 RTP RTP:PayloadType = PCMU Audio, 8000Hz [1 Channel], SSRC = 242919752, Seq = 1, TimeStamp = 160 {UDP:38, IPv4:15}
53 1:35:00 PM 5/2/2013 5.6131312 10.0.71.24 10.0.71.109 RTP RTP:PayloadType = PCMU Audio, 8000Hz [1 Channel], SSRC = 242919752, Seq = 2, TimeStamp = 320 {UDP:38, IPv4:15}
54 1:35:00 PM 5/2/2013 5.6452028 10.0.71.24 10.0.71.109 RTP RTP:PayloadType = PCMU Audio, 8000Hz [1 Channel], SSRC = 242919752, Seq = 3, TimeStamp = 480 {UDP:38, IPv4:15}
55 1:35:01 PM 5/2/2013 5.6729429 10.0.71.24 10.0.71.109 RTP RTP:PayloadType = PCMU Audio, 8000Hz [1 Channel], SSRC = 242919752, Seq = 4, TimeStamp = 640 {UDP:38, IPv4:15}
56 1:35:01 PM 5/2/2013 5.6729429 10.0.71.24 10.0.71.109 RTP RTP:PayloadType = PCMU Audio, 8000Hz [1 Channel], SSRC = 242919752, Seq = 5, TimeStamp = 800 {UDP:38, IPv4:15}
58 1:35:01 PM 5/2/2013 5.7052129 10.0.71.24 10.0.71.109 RTP RTP:PayloadType = PCMU Audio, 8000Hz [1 Channel], SSRC = 242919752, Seq = 6, TimeStamp = 960 {UDP:38, IPv4:15}
59 1:35:01 PM 5/2/2013 5.7330918 10.0.71.24 10.0.71.109 RTP RTP:PayloadType = PCMU Audio, 8000Hz [1 Channel], SSRC = 242919752, Seq = 7, TimeStamp = 1120 {UDP:38, IPv4:15}
60 1:35:01 PM 5/2/2013 5.7331906 10.0.71.24 10.0.71.109 RTP RTP:PayloadType = PCMU Audio, 8000Hz [1 Channel], SSRC = 242919752, Seq = 8, TimeStamp = 1280 {UDP:38, IPv4:15}
61 1:35:01 PM 5/2/2013 5.7650426 10.0.71.24 10.0.71.109 RTP RTP:PayloadType = PCMU Audio, 8000Hz [1 Channel], SSRC = 242919752, Seq = 9, TimeStamp = 1440 {UDP:38, IPv4:15}
62 1:35:01 PM 5/2/2013 5.8048753 10.0.71.24 10.0.71.109 RTP RTP:PayloadType = PCMU Audio, 8000Hz [1 Channel], SSRC = 242919752, Seq = 10, TimeStamp = 1600 {UDP:38, IPv4:15}
63 1:35:01 PM 5/2/2013 5.8048753 10.0.71.24 10.0.71.109 RTP RTP:PayloadType = PCMU Audio, 8000Hz [1 Channel], SSRC = 242919752, Seq = 11, TimeStamp = 1760 {UDP:38, IPv4:15}
64 1:35:01 PM 5/2/2013 5.8327720 10.0.71.24 10.0.71.109 RTP RTP:PayloadType = PCMU Audio, 8000Hz [1 Channel], SSRC = 242919752, Seq = 12, TimeStamp = 1920 {UDP:38, IPv4:15}
65 1:35:01 PM 5/2/2013 5.8328593 10.0.71.24 10.0.71.109 RTP RTP:PayloadType = PCMU Audio, 8000Hz [1 Channel], SSRC = 242919752, Seq = 13, TimeStamp = 2080 {UDP:38, IPv4:15}
[snip]
164 1:35:02 PM 5/2/2013 7.5641524 10.0.71.109 10.0.71.24 RTCP RTCP:RTCP compound packet - Number of packets = 0x2 {UDP:49, IPv4:15}
==== END: 10.0.71.109 (playSIP host) network log ============================================================
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