[Home]

Summary:ASTERISK-20464: Can't join ConfBridge() with video
Reporter:Leif Madsen (lmadsen)Labels:
Date Opened:2012-09-24 08:53:31Date Closed:2012-09-24 09:27:19
Priority:MajorRegression?Yes
Status:Closed/CompleteComponents:Applications/app_confbridge
Versions:11.0.0-beta2 Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) permissive-h264.diff
Description:When attempting to join a conference bridge (ConfBridge()) with video support from jitsi, the call is rejected by jitsi saying the session description is invalid.

It appears this line in the 200 OK is invalid:

a=rtpmap:96 /0
Comments:By: Leif Madsen (lmadsen) 2012-09-24 08:54:39.503-0500

{noformat}
<--- SIP read from UDP:172.16.0.195:5060 --->
INVITE sip:602@172.16.0.152 SIP/2.0
Call-ID: c831bdc03096733a2196500c44f6ef65@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
From: "0000FFFF0001" <sip:0000FFFF0001@172.16.0.152>;tag=41600faa
To: <sip:602@172.16.0.152>
Max-Forwards: 70
Contact: "0000FFFF0001" <sip:0000FFFF0001@172.16.0.195:5060;transport=udp;registering_acc=172_16_0_152>
User-Agent: Jitsi1.1-nightly.4211Linux
Content-Type: application/sdp
Via: SIP/2.0/UDP 172.16.0.195:5060;branch=z9hG4bK-373734-d1fabe15891a389e883a0a67a7e20385
Authorization: Digest username="0000FFFF0001",realm="asterisk",nonce="719ed936",uri="sip:602@172.16.0.152",response="f0972e8bc0dfe0648c08af313b267859",algorithm=MD5
Content-Length: 742

v=0
o=0000FFFF0001 0 0 IN IP4 172.16.0.195
s=-
c=IN IP4 172.16.0.195
t=0 0
m=audio 5032 RTP/AVP 9 0 8 3
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level
a=zrtp-hash:1.10 dbee20aa6a083042dad6b51c000e1fa4451dd578c2849a8f0e858377d823e2d5
m=video 5034 RTP/AVP 96 99
a=recvonly
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=4DE01f;packetization-mode=1
a=imageattr:96 send [x=[0-160],y=[0-100]] recv [x=[0-1920],y=[0-1080]]
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=4DE01f
a=imageattr:99 send [x=[0-160],y=[0-100]] recv [x=[0-1920],y=[0-1080]]
a=zrtp-hash:1.10 05b26541603785fb2b26cc19bf66c35fe1bbeaab533a54ed2a9437819b20e866
<------------->
--- (12 headers 21 lines) ---
Sending to 172.16.0.195:5060 (no NAT)
Using INVITE request as basis request - c831bdc03096733a2196500c44f6ef65@0:0:0:0:0:0:0:0
Found peer '0000FFFF0001' for '0000FFFF0001' from 172.16.0.195:5060
 == Using SIP RTP CoS mark 5
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found RTP video format 96
Found RTP video format 99
Found video description format H264 for ID 96
Found video description format H264 for ID 99
Capabilities: us - (gsm|ulaw|alaw|g722|h264), peer - audio=(gsm|ulaw|alaw|g722)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 172.16.0.195:5032
Peer video RTP is at port 172.16.0.195:5034
Looking for 602 in LocalSets (domain 172.16.0.152)
list_route: hop: <sip:0000FFFF0001@172.16.0.195:5060;transport=udp;registering_acc=172_16_0_152>





----- snip ------



   -- Executing [conference@LocalSets:3] ConfBridge("SIP/0000FFFF0001-00000024", "primary,,,volume_ctrl_menu") in new stack
Audio is at 12964
Video is at 172.16.0.152:21528
Adding codec 100012 (g722) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding video codec 200004 (unknown) to SDP

<--- Reliably Transmitting (no NAT) to 172.16.0.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.195:5060;branch=z9hG4bK-373734-d1fabe15891a389e883a0a67a7e20385;received=172.16.0.195
From: "0000FFFF0001" <sip:0000FFFF0001@172.16.0.152>;tag=41600faa
To: <sip:602@172.16.0.152>;tag=as033212ea
Call-ID: c831bdc03096733a2196500c44f6ef65@0:0:0:0:0:0:0:0
CSeq: 2 INVITE
Server: Asterisk PBX SVN-trunk-r372808
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:602@172.16.0.152:5060>
Content-Type: application/sdp
Content-Length: 381

v=0
o=root 1575016428 1575016428 IN IP4 172.16.0.152
s=Asterisk PBX SVN-trunk-r372808
c=IN IP4 172.16.0.152
b=CT:384
t=0 0
m=audio 12964 RTP/AVP 9 0 8 3
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=ptime:20
a=sendrecv
m=video 21528 RTP/AVP 96
a=rtpmap:96 /0
a=fmtp:96 profile-level-id=4DE01F;packetization-mode=1
a=sendrecv
{noformat}
By: Joshua C. Colp (jcolp) 2012-09-24 09:27:19.607-0500

Fixed in 11 as of revision 373413 and trunk as of revision 373414.