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Summary:ASTERISK-20315: incomplete SDP when using websocket transport
Reporter:Roman Yakovskiy (zerr0)Labels:
Date Opened:2012-08-24 03:36:24Date Closed:2012-08-24 08:23:17
Priority:MajorRegression?
Status:Closed/CompleteComponents:Channels/chan_sip/Video Channels/chan_sip/WebSocket
Versions:11.0.0-beta1 Frequency of
Occurrence
Constant
Related
Issues:
Environment:FreeBSD 9.0; client - sipML5, chrome 21.0.1180.57Attachments:
Description:using transport=ws, making a call. get 183 Session Progress from asterisk with SDP.
SDP is incomplete. no info about videostream.

SDP looks like this:
v=0
o=root 508220761 508220761 IN IP4 192.168.0.247
s=Asterisk PBX SVN-trunk-r371633
c=IN IP4 192.168.0.247
t=0 0
m=audio 11710 RTP/SAVPF 0 0 8 8 102 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=ice-ufrag:090dca517e974c5d605c09dc0006dea2
a=ice-pwd:6e80a9f24c63ee88672c6110224fd680
a=candidate:Hc0a800f7 1 UDP 2130706431 192.168.0.247 11710 typ host
a=candidate:Hc0a800f7 2 UDP 2130706430 192.168.0.247 11711 typ host
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:4qW/Tu1Llj/o5iKs95iiqMqlkksofe/JDkQZrAUQ
m=video 0 RTP/SAVPF 100 101 102
Comments:By: Joshua C. Colp (jcolp) 2012-08-24 05:17:20.547-0500

The video stream is way at the bottom. The 0 as the port means that Asterisk did not accept it for whatever reason. Could be codec related. This is proper SDP.

By: Matt Jordan (mjordan) 2012-08-24 08:23:17.884-0500

Closing as not a bug.  If you can provide additional information demonstrating that Asterisk should have accepted the video stream, we can revisit this issue.  Please contact a bug marshal in #asterisk-bugs if you believe you have such information.