[Asterisk-video] Branches Videocaps

[Valerio Puglia]

yes it's work on 1.6 beta 7.1 with the same problem i describe below:

i try to use videcaps for settings a videocall bandwith and other option..

I have made a mistake:
the kapanga works but the video is terrible.... in incoming calls but is 
perfect on videcall out..
the sdp that kapanga send to outgoing calls to asterisk is

o=9001 1208967526 1208968601 IN IP4 10.0.2.182
s=Kapanga [1208967526]
c=IN IP4 10.0.2.182
t=0 0
m=audio 5130 RTP/AVP 0 8 101
a=rtpmap:0 pcmu/8000
a=sendrecv
a=silenceSupp:off - - -
a=rtcp:5131
a=maxptime:20
a=ptime:20
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,36
m=video 5132 RTP/AVP 98 105
a=rtpmap:98 theora/90000
a=fmtp:98 sampling=YCbCr-4:2:0; width=176, height=144;
b=AS:61
a=sendrecv
a=rtcp:5133
a=rtpmap:105 h263-1998/90000
a=fmtp:105 profile=0; level=10

on videocall out kapanga set the bandwith information with b=AS:xx
and video works very well
but on sip.conf is possible to set only b=CT:XXX with the option 
maxcallbitrate=xxx
is possible to set maxvideobitrate (b=AS:xx)?

because i think it is caused by that (the video bitrate too high)




Klaus Darilion ha scritto:
> 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?
>>
>>     
>
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-- 
Best regards,
Valerio Puglia
OScorp S.P.A.
NETWORK Adm