Elastix Tandberg Video support H.264

gogasca

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#1
I have 2 Tandberg E20s that video works between each other but...
quality is negotiated to lower one (QCIF).
I see that Elastix removes fmtp from SDP messages

768 x 448@30fps (w448p)
576 x 448@30fps (448p)
512 x 288@30fps (w288p)
352 x 288@30fps (CIF)
176 x 144@30fps (QCIF)

I see that when we get the INVITE with SDP Elastix removes fmtp values when forwarding INVITE to other peer.

INVITE sip:103@110.10.0.200 SIP/2.0
Via: SIP/2.0/UDP 110.10.0.144:5060;branch=z9hG4bK90b3d0e51cfdbe6e4602a611dcfd5918.1;rport
Call-ID: 201764a6216109dd@110.10.0.144
CSeq: 101 INVITE
Contact: <sip:104@110.10.0.144:5060>
From: <sip:104@110.10.0.200>;tag=5420a486a80dbc77
To: <sip:103@110.10.0.200>
Max-Forwards: 70
Route: <sip:110.10.0.200;lr>
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,INFO,OPTIONS,REFER,NOTIFY
User-Agent: TANDBERG/257 (TE2.2.1.224666)
Authorization: Digest nonce="2b87ef6a", realm="asterisk", username="104", uri="sip:110.10.0.200", response="f2c6901e89bf6825116c03e4822c7009", algorithm=MD5
Supported: replaces,100rel,timer,gruu,path,outbound
Session-Expires: 500
Content-Type: application/sdp
Content-Length: 1225

v=0
o=tandberg 4 1 IN IP4 110.10.0.144
s=-
c=IN IP4 110.10.0.144
b=CT:768
t=0 0
m=audio 2334 RTP/AVP 100 101 102 9 18 11 8 0 103
c=IN IP4 110.10.0.144
b=TIAS:64000
a=rtpmap:100 MP4A-LATM/90000
a=fmtp:100 profile-level-id=24;object=23;bitrate=64000
a=rtpmap:101 G7221/16000
a=fmtp:101 bitrate=32000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=24000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:11 L16/16000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:103 telephone-event/8000
a=fmtp:103 0-15
a=sendrecv
m=video 2336 RTP/AVP 97 98 99 34 31
b=TIAS:768000
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42800d;max-mbps=40500;max-fs=1344;max-smbps=40500
a=rtpmap:98 H264/90000
a=fmtp:98 profile-level-id=42800d;max-mbps=40500;max-fs=1344;max-smbps=40500;packetization-mode=1
a=rtpmap:99 H263-1998/90000
a=fmtp:99 custom=1024,768,4;custom=1024,576,4;custom=800,600,4;cif4=2;custom=720,480,2;custom=640,480,2;custom=512,288,1;cif=1;custom=352,240,1;qcif=1;maxbr=7680
a=rtpmap:34 H263/90000
a=fmtp:34 cif=1;qcif=1;maxbr=7680
a=rtpmap:31 H261/90000
a=fmtp:31 cif=1;qcif=1;maxbr=7680
a=rtcp-fb:* nack pli
a=sendrecv
a=content:main
a=label:11
a=answer:full


Found RTP video format 97
Found RTP video format 98
Found RTP video format 99
Found RTP video format 34
Found RTP video format 31
Found video description format H264 for ID 97
Found video description format H264 for ID 98
Found video description format H263-1998 for ID 99
Found video description format H263 for ID 34
Found video description format H261 for ID 31

===

Audio is at 110.10.0.200 port 17922
Video is at 110.10.0.200 port 14696
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x1000 (g722) to SDP
Adding video codec 0x200000 (h264) to SDP
Adding non-codec 0x1 (telephone-event) to SDP


Reliably Transmitting (NAT) to 110.10.0.136:5060:
INVITE sip:103@110.10.0.136:5060 SIP/2.0
Via: SIP/2.0/UDP 110.10.0.200:5060;branch=z9hG4bK3f194b88;rport
Max-Forwards: 70
From: "Mr T" <sip:104@110.10.0.200>;tag=as21c38e6e
To: <sip:103@110.10.0.136:5060>
Contact: <sip:104@110.10.0.200>
Call-ID: 456556eb59a5c74b57ccb1461e79ead1@110.10.0.200
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.10
Date: Sat, 20 Nov 2010 06:15:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 330

v=0
o=root 29229991 29229991 IN IP4 110.10.0.200
s=Asterisk PBX 1.6.2.10
c=IN IP4 110.10.0.200
b=CT:1080
t=0 0
m=audio 17922 RTP/AVP 0 9 103
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:103 telephone-event/8000
a=fmtp:103 0-16
a=ptime:20
a=sendrecv
m=video 14696 RTP/AVP 97
a=rtpmap:97 H264/90000
a=sendrecv


Any idea?
How can modify chan_sip.c ?

http://lists.digium.com/pipermail/aster ... 02727.html

Thanks
 

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