[Asterisk-Users] Video calls (MS Messenger, Tandberg)

[Jens.Kammann at dlr.de]

Hi,

According to http://www.voip-info.org/wiki-Asterisk+video it should be
possible to place video calls using asterisk.
So far I managed to get both Microsoft Messenger and a video conference
system from Tandberg to register with asterisk. Voice calls between both
stations work perfectly (using ulaw codec).

Video calls fail with asterisk putting the tandberg system "on hold"
(playing Music-on-hold).

Despite both clients claim "H261/H263" codecs, SDP negotiation results:

> Capabilities: us - 0xc020e(GSM|ULAW|ALAW|SPEEX|H261|H263), peer -
audio=0xe(GSM|ULAW|ALAW)/video 0x0(EMPTY), combined - 0xe(GSM|ULAW|ALAW)
>Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined -
0x1(G723)

So no common video codec was negotiated (thus connections is voice-only)

Any ideas? Do I need to active the H261/H263 codecs somewhere? I tried
forcing theses codecs in sip.conf, but no luck either.

regards,
   Jens 

SIP/SDP Debug for 


Sip read: 
INVITE sip:52800 SIP/2.0
Via: SIP/2.0/UDP 129.247.XXX.XXX:5060;branch=z9hG4bK2375534000-17264122
Max-Forwards: 70
From:
59999<sip:59999 at 129.247.XXX.XXX>;epid=82052805FAC6AK;tag=plcm_2375520000
-17264121
To: <sip:52800>
Call-ID: 2375519000-17264119
CSeq: 2 INVITE
Session-Expires: 90
Supported: timer
Contact: <sip:129.247.XXX.XXX:5060;transport=udp>
Content-Type: application/sdp
Proxy-Authorization: Digest
username="59999",realm="asterisk",nonce="6f505027",uri="sip:52800",respo
nse="d260786708039bed1a05af94a4a69fb3",algorithm=md5
User-Agent: Polycom VSX 7000A Release 8.0.3 - 06Oct2005 13:49
Content-Length: 990

v=0
o=DLR-KN 1353514857 0 IN IP4 129.247.173.207
s=-
c=IN IP4 129.247.XXX.XXX
b=AS:384
t=0 0
m=audio 49184 RTP/AVP 99 98 97 102 101 103 9 15 0 8 18
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=rtpmap:15 G728/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729A/8000
a=fmtp:18 annexb=no
a=sendrecv
m=video 49186 RTP/AVP 109 34 96 31
b=TIAS:384000
a=rtpmap:109 H264/90000
a=fmtp:109 profile-level-id=42800c; max-mbps=10000; max-fs=1792;
max-br=775
a=rtpmap:34 H263/90000
a=rtpmap:96 H263-1998/90000
a=fmtp:96 CIF4=2;CIF=1;QCIF=1;SQCIF=1;F;J;T
a=rtp
14 headers, 34 lines
Using latest request as basis request
Sending to 129.247.XXX.XXX : 5060 (NAT)
Found RTP audio format 99
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 102
Found RTP audio format 101
Found RTP audio format 103
Found RTP audio format 9
Found RTP audio format 15
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Peer audio RTP is at port 129.247.XXX.XXX:49184
Found 
description format SIREN14
Found description format SIREN14
Found description format SIREN14
Found description format G7221
Found description format G7221
Found description format G7221
Found description format G722
Found description format G728
Found description format PCMU
Found description format PCMA
Found description format G729A
Found description format H264
Found description format H263
Found description format H263-1998
Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer -
audio=0x50c(ULAW|ALAW|G729A|ILBC)/video=0x0(EMPTY), combined -
0xc(ULAW|ALAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined -
0x1(G723)
Found user '59999'
Looking for 52800 in sip_dlrpbx
list_route: hop: <sip:129.247.XXX.XXX:5060;transport=udp>
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
129.247.XXX.XXX:5060;branch=z9hG4bK2375534000-17264122;received=129.247.
XXX.XXX;rport=5060
From:
59999<sip:59999 at 129.247.XXX.XXX>;epid=82052805FAC6AK;tag=plcm_2375520000
-17264121
To: <sip:52800>;tag=as7a4b0ce2
Call-ID: 2375519000-17264119
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:52800 at 129.247.XXX.XXX>
Content-Length: 0