20 jun 2006 kl. 14.28 skrev Mindaugas Kuprys:
Video started to work. Now intresting thing is that video size is
half reduced than calling directly from phone to phone. Phones:
Tatung tia-8800. I have attached sip messages. that else might be
important..? one of phones is behind nat.
The reason for the half size is that we do not support the fmtp:
header yet. If you want to join in the work with enhancing
the video support we have a developer's list at asterisk-video, see
http://lists.digium.com
Thanks for the debug, it helps. In svn trunk, we have improved video
support a bit, so that Asterisk won't offer
another video codec on the outbound call like in this dialog,
considering we don't transcode video.
/Olle
Btw: The following packet indicates to me that the SIP firmware in
this phone is rather buggy:
U 217.9.241.234:5060 -> 217.9.241.227:5060
SIP/2.0 200 OK.
From: "Giedrius"<sip:[EMAIL PROTECTED]>;tag=as009e18fc.
To: <sip:[EMAIL PROTECTED]:5060>;tag=0-13c4-7fa-1f0bc6-74fe10e7.
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE.
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS.
User-Agent: TTIC Video phone.
Via: SIP/2.0/UDP 217.9.241.227:5060;rport=5060;branch=z9hG4bK198aadad.
Contact: <sip:[EMAIL PROTECTED]:5060>.
Content-Type: application/SDP.
Content-Length: 723.
.
v=0.
o=Server 2890844527 2890844527 IN IP4 217.9.241.234.
s=session.
c=IN IP4 217.9.241.234.
t=0 0.
m=audio 6070 RTP/AVP 0 4 97 8 18 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0.
a=rtpmap:0 PCMU/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:97 telephone-event/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:0 PCMU/8000.
m=video 6072 RTP/AVP 34.
b=AS:768.
a=fmtp:34 CIF=2 QCIF=1 MaxBR=7680.
a=sendrecv.
a=rtpmap:34 H263/90000.
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