Hello!
I recently upgraded the Kubuntu Linux version to 11.10, which upgraded Asterisk from version 1.6.2.9-2 to 1.8.4.4.
With version 1.6.2.9-2, video worked fine. With version 1.8.4.4, most of the times, Asterisk does not send a video RTP stream.
The configuration is simple: under sip.conf uncomment:
videosupport=on
Using X-Lite, and calling the “demo” 600 echo test extension, I get the video echoed back. So far, so good.
However, if I call Voicemail at 1235, record a message with video, the .h263 file is recorded along with the .gsm and .wav files. But if I call VoicemailMain at extension 8500 and listen to the messages, I get no video, just the audio. Using the Playback function in a script only reproduces the audio, even though the video files exist.
It worked before. Is there any additional configuration needed under Asterisk version 1.8.4.4?
I am using 2 PCs connected through a LAN. Using Wireshar, I see that X-Lite sends the video, but Asterisk does not.
The “INVITE” and “200 Ok” messages follow with the verbose CLI output. All 3 messages in the voicemail have video. msg0000 and msg0001 were recorded with Asterisk v1.6.2.9-2 and msg0002 was recorded with Asterisk v1.8.4.4. It is curious to notice that different extensions are shown in the CLI output when playing them (.gsm and .slin), according to the Asterisk version used in the recording.
Thanks,
Paulo
INVITE sip:8500@mia.inesc-id.pt SIP/2.0
Via: SIP/2.0/UDP 146.193.48.14:31860;branch=z9hG4bK-d87543-fe71951e8733967e-1–d87543-;rport
Max-Forwards: 70
Contact: sip:4000@146.193.48.14:31860
To: "8500"sip:8500@mia.inesc-id.pt
From: "Paulo Rogério Pereira"sip:4000@mia.inesc-id.pt;tag=af63c829
Call-ID: NmVhYWVjMGY1OGEzNmUyYjA3OTA4MjNiZjUwODA2Nzc.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 841
v=0
o=- 6 2 IN IP4 146.193.48.14
s=CounterPath X-Lite 3.0
c=IN IP4 146.193.48.14
t=0 0
m=audio 1940 RTP/AVP 107 119 100 106 0 105 98 8 101
a=alt:1 3 : KQJbRoA7 Xnwe9/N6 146.193.48.14 1940
a=alt:2 2 : 1Pxo3NhN wierv6fz 192.168.200.1 1940
a=alt:3 1 : wPZ1P7ah xcaxcBeH 192.168.152.1 1940
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
m=video 28760 RTP/AVP 115 34
a=alt:1 3 : v8yWK+EQ We6olgA8 146.193.48.14 28760
a=alt:2 2 : 37l/vV6O OmEbgyI6 192.168.200.1 28760
a=alt:3 1 : jN6VEi1a BRUuVq5c 192.168.152.1 28760
a=fmtp:115 QCIF=1 I=1 J=1 T=1 MaxBR=1960
a=fmtp:34 QCIF=1 MaxBR=1960
a=rtpmap:115 H263-1998/90000
a=rtpmap:34 H263/90000
a=sendrecv
SIP/2.0 200 OK
Via: SIP/2.0/UDP 146.193.48.14:31860;branch=z9hG4bK-d87543-fe71951e8733967e-1–d87543-;received=146.193.48.14;rport=31860
From: "Paulo Rogério Pereira"sip:4000@mia.inesc-id.pt;tag=af63c829
To: "8500"sip:8500@mia.inesc-id.pt;tag=as5aca65e7
Call-ID: NmVhYWVjMGY1OGEzNmUyYjA3OTA4MjNiZjUwODA2Nzc.
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.4.4~dfsg-2ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:8500@146.193.48.17:5060
Content-Type: application/sdp
Content-Length: 348
v=0
o=root 1380058181 1380058181 IN IP4 146.193.48.17
s=Asterisk PBX 1.8.4.4~dfsg-2ubuntu1
c=IN IP4 146.193.48.17
b=CT:384
t=0 0
m=audio 16204 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 19116 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
Asterisk console:
Connected to Asterisk 1.8.4.4~dfsg-2ubuntu1 currently running on mia (pid = 1160)
Verbosity was 0 and is now 17
== Using SIP RTP CoS mark 5
– Executing [8500@demo:1] VoiceMailMain(“SIP/4000-00000002”, “”) in new stack
– <SIP/4000-00000002> Playing ‘vm-login.gsm’ (language ‘en’)
[Nov 28 11:42:47] NOTICE[4191]: channel.c:4070 __ast_read: Dropping incompatible voice frame on SIP/4000-00000002 of format alaw since our native format has changed to 0x180004 (ulaw|h263|h263p)
[Nov 28 11:42:47] NOTICE[4191]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from ‘146.193.48.14:7900’
[Nov 28 11:42:47] NOTICE[4191]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from ‘146.193.48.14:7900’
[Nov 28 11:42:47] NOTICE[4191]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from ‘146.193.48.14:7900’
[Nov 28 11:42:55] NOTICE[1190]: chan_sip.c:23613 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 4000
– <SIP/4000-00000002> Playing ‘vm-password.gsm’ (language ‘en’)
[Nov 28 11:42:58] NOTICE[4191]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from ‘146.193.48.14:7900’
– <SIP/4000-00000002> Playing ‘vm-youhave.gsm’ (language ‘en’)
– <SIP/4000-00000002> Playing ‘digits/3.gsm’ (language ‘en’)
– <SIP/4000-00000002> Playing ‘vm-Old.gsm’ (language ‘en’)
– <SIP/4000-00000002> Playing ‘vm-messages.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘vm-first.gsm’ (language ‘en’)
== Parsing ‘/var/spool/asterisk/voicemail/default/1234/Old/msg0000.txt’: == Found
– <SIP/4000-00000004> Playing ‘vm-message.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘vm-received.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘digits/mon-3.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘digits/h-8.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘digits/2.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘digits/thousand.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘digits/11.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘digits/at.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘digits/12.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘digits/40.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘digits/5.gsm’ (language ‘en’)
[Nov 28 11:44:54] NOTICE[4256]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from ‘146.193.48.14:48308’
– <SIP/4000-00000004> Playing ‘digits/p-m.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘/var/spool/asterisk/voicemail/default/1234/Old/msg0000.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘vm-advopts.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘vm-repeat.gsm’ (language ‘en’)
== Parsing ‘/var/spool/asterisk/voicemail/default/1234/Old/msg0001.txt’: == Found
– <SIP/4000-00000004> Playing ‘vm-message.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘digits/2.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘vm-received.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘digits/mon-3.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘digits/h-8.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘digits/2.gsm’ (language ‘en’)
[Nov 28 11:45:05] NOTICE[4256]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from ‘146.193.48.14:48308’
– <SIP/4000-00000004> Playing ‘digits/thousand.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘digits/11.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘digits/at.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘digits/12.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘digits/40.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘digits/4.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘digits/p-m.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘/var/spool/asterisk/voicemail/default/1234/Old/msg0001.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘vm-prev.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘vm-last.gsm’ (language ‘en’)
[Nov 28 11:45:15] NOTICE[4256]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from ‘146.193.48.14:48308’
== Parsing ‘/var/spool/asterisk/voicemail/default/1234/Old/msg0002.txt’: == Found
– <SIP/4000-00000004> Playing ‘vm-message.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘vm-received.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘digits/day-5.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘digits/at.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘digits/4.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘digits/oh.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘digits/1.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘digits/p-m.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘/var/spool/asterisk/voicemail/default/1234/Old/msg0002.slin’ (language ‘en’)
[Nov 28 11:45:26] NOTICE[4256]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126 received from ‘146.193.48.14:48308’
– <SIP/4000-00000004> Playing ‘vm-prev.gsm’ (language ‘en’)
– <SIP/4000-00000004> Playing ‘vm-advopts.gsm’ (language ‘en’)
== Spawn extension (demo, 8500, 1) exited non-zero on ‘SIP/4000-00000004’
[Nov 28 11:45:55] NOTICE[1190]: chan_sip.c:23613 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 4000