[asterisk-users] [SIP/H.264] Codec negotiation problem ?

[Nicolas Bourbaki]

Hi,

I've a problem configuring my Asterisk. What I try to reach is to
interconnect a Tandberg Visioconference (SIP) world with my Asterisk (SIP)
with 1 constraint I can't change : "every RTP flow needs to pass THROUGH
Asterisk, and are NOT nated"

What I observe :
  - a call made from a SIP Phone registred in Asterisk to Tandberg works
(voice and video bidirectionnal)
  - a call made from to Tandberg to the SIP phone doesn't work (voice
bidirectionnal, voice only received by the SIP phone, no incomming video for
Tandberg)

I think the problem may  come from codec negotiotation :
  - when call is made from the SIP phone, it uses "code" 99 for H.264 codec,
as Asterisk. Tdb reply SIP:Ok with the same number for H.264
  - when call is made from Tbd, it uses "code" 98 for H.264 codec. Asterisk
then send the Invite with 99 as codec number


I use the version 1.6.2.6 of Asterisk

Is this kind of configuration supposed to work ? I know passing video media
through Asterisk may not be optimal, but I really need it, even if I have to
patch Asterisk

Thanks for your help



SDP send by Tandberg :
--------------------------
v=0
o=tandberg 1 5 IN IP4 192.168.50.10
s=-
c=IN IP4 192.168.50.10
b=CT:1920
t=0 0
m=audio 48260 RTP/AVP 100 101 9 8 0 102
b=TIAS:64000
a=rtpmap:100 G7221/16000
a=fmtp:100 bitrate=32000
a=rtpmap:101 G7221/16000
a=fmtp:101 bitrate=24000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-15
a=sendrecv
m=video 48262 RTP/AVP 97 98 99 34 31
c=IN IP4 192.168.50.10
b=TIAS:1920000
a=rtpmap:97 H264-RCDO/90000
a=fmtp:97 profile-level-id=008016;max-
mbps=42000;max-fs=3600;max-smbps=323500
a=rtpmap:98 H264/90000
a=fmtp:98
profile-level-id=428016;max-mbps=35000;max-fs=3600;max-smbps=323500
a=rtpmap:99 H263-1998/90000
a=fmtp:99
custom=1280,800,0;custom=1280,768,0;custom=1280,720,3;custom=1024,768,4;custom=1024,576,2;custom=800,600,3;cif4=2;custo
a=rtpmap:34 H263/90000
a=fmtp:34 cif4=2;cif=1;qcif=1;sqcif=1;maxbr=19200
a=rtpmap:31 H261/90000
a=fmtp:31 cif=1;qcif=1;maxbr=19200
a=rtcp-fb:* nack pli
a=sendrecv
a=content:main
a=label:11
a=answer:full
m=application 5078 UDP/BFCP *
c=IN IP4 192.168.50.10
a=floorctrl:c-s
a=confid:1
a=floorid:2 mstrm:12
a=userid:1
a=setup:passive
a=connection:new
m=video 48264 RTP/AVP 99 34 31
c=IN IP4 192.168.50.10
b=TIAS:1920000
a=rtpmap:99 H263-1998/90000
a=fmtp:99
custom=1280,800,0;custom=1280,768,0;custom=1280,720,3;custom=1024,768,4;custom=1024,576,2;custom=800,600,3;cif4=2;custo
a=rtpmap:34 H263/90000
a=fmtp:34 cif4=2;cif=1;qcif=1;sqcif=1;maxbr=19200
a=rtpmap:31 H261/90000
a=fmtp:31 cif=1;qcif=1;maxbr=19200
a=rtcp-fb:* nack pli
a=sendrecv
a=content:slides
a=label:12



SDP send by Asterisk
v=0
o=root 1077353049 1077353049 IN IP4 192.168.13.100
s=Asterisk PBX 1.6.2.6
c=IN IP4 192.168.13.100
b=CT:384
t=0 0
m=audio 14604 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 17962 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendrecv





Here is my sip.conf
--------------------
[general]
port = 5060
bindaddr = 0.0.0.0
disallow = all
realm = testRealm

allow = ulaw
allow = h264
videosupport=yes

canreinvite = no
calleridupdate = info
usercallerid = no
context = default

[toTandberg]
host=192.168.50.53
type=friend
qualify=yes
qualifyreq=1
------------------
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