[Asterisk-video] Branches Videocaps

[Klaus Darilion]

does it work with plain Asterisk (not videocaps branch)?

regards
klaus

Valerio Puglia schrieb:
> I try to use this branches with fontventa application
> 
> the version is svn 114578
> 
> when the call is redirect on siphone eyeBeam 1.5 or kapanga the video 
> sent by cellular doesn't start...on x lite give this erorr "SIP/2.0 415 
> Unsupported Media Type"
> 
> the sdp of invite is
> 
> v=0
> o=root xxxx
> s=Asterisk PBX SVN-oej-videocaps-r114506M-/trunk
> c=IN IP4 x.x.x.x
> b=CT:6000
> t=0 0
> m=audio 11428 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> m=video 10488 RTP/AVP 34 98
> a=rtpmap:34 H263/90000
> a=fmtp:34;CIF=1;QCIF=1;maxbr=3840
> a=rtpmap:98 h263-1998/90000
> a=fmtp:98;CIF=1;QCIF=1;maxbr=3840
> a=sendrecv
> 
> 
> 
> CSeq: 102 INVITE
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, 
> SUBSCRIBE, INFO
> Content-Type: application/sdp
> User-Agent: eyeBeam release 1100l stamp 46320
> Content-Length: 250
> 
> v=0
> o=- 1 2 IN IP4 10.0.2.5
> s=CounterPath eyeBeam 1.5
> c=IN IP4 62.94.144.245
> t=0 0
> m=audio 37208 RTP/AVP 0 8 101
> a=fmtp:101 0-15
> a=rtpmap:101 telephone-event/8000
> a=sendrecv
> m=video 44800 RTP/AVP 98
> a=rtpmap:98 H263-1998/90000
> a=sendrecv
> 
> i try to change the parameter of video codec on sip.conf to
> 
> h263=cif=30 qcif=30 maxbr=1024
> h263p=cif=30 qcif=30 maxbr=1024
> 
> but in sdp nothing has changed "a=fmtp:98;CIF=1;QCIF=1;maxbr=3840"
> 
> on outgoing call the video on sipphone is ok
> 
> any idea?
>