[asterisk-users] Headaches with Video over SIP

Peter Howard peter.howard at ursys.com.au
Sun Nov 12 23:11:27 MST 2006


On Mon, 2006-11-13 at 13:42 +0800, Rosli Sukri wrote:
> any logs/errors when you do a verbose 6 and a sip debug ?
> 

I've been running with verbose 9, debug 9, and sip debug.  The resultant
output seems fine. The only warning is:

WARNING[6964]: chan_sip.c:3592 process_sdp: Unknown SDP media type in
offer: data 49218 RTP/AVP 100

The rest of the output seems to be normal.  I can regenerate it, but
right now I've put 1.4-beta3 on to see if that improves things (so far
it hasn't, but I've tried one run)


> On 11/13/06, Peter Howard <peter.howard at ursys.com.au> wrote:
>         On Mon, 2006-11-13 at 00:57 +0100, Patrick wrote:
>         > On Mon, 2006-11-13 at 10:45 +1100, Peter Howard wrote: 
>         > > Greetings all,
>         > >
>         > > I'm playing with asterisk and two Polycom VSX300
>         videoconferencing
>         > > units.  And I'm having zero luck getting video working
>         over SIP.
>         > >
>         > > The two units register fine with asterisk, and with
>         "allow=all" in 
>         > > sip.conf, the two units establish voice.  But no
>         video.  And no obvious
>         > > messages as to whats going wrong.  The config for each is
>         (they're
>         > > numbered 201 and 202):
>         > >
>         > > [202] 
>         > > secret=
>         > > type=friend
>         > > context=from-sip-202
>         > > host=dynamic
>         > > nat=no
>         > > canreinvite=yes
>         > > dtmfmode=rfc2833
>         > > disallow=all
>         > > allow=all 
>         > >
>         > >
>         > > If you're wondering why I do the "disallow=all"
>         immediately followed by
>         > > "allow=all", it's because the allow line has spent a lot
>         of time with
>         > > restricted codecs to see if that makes a difference. 
>         > >
>         > > I can provide the full sip.conf, extensions.conf, and
>         debug output if
>         > > anyone wants to see them.
>         > >
>         > > Any suggestions as to where things are falling down?
>         > 
>         > Do you have "videosupport=yes" in your sip.conf?
>         
>         Yes I do.  I've also confirmed that I have a version of
>         asterisk which
>         includes the patch for H263P (which is what the Polycoms want
>         to talk).
>         
>         --
>         Peter Howard
>         URSYS
>         13 Burwood Rd,
>         Burwood, NSW 2134
>         
>         Ph: 02 8745 2816    Fax: 02 8745 2828
>         
>         _______________________________________________
>         --Bandwidth and Colocation provided by Easynews.com --
>         
>         asterisk-users mailing list
>         To UNSUBSCRIBE or update options visit:
>            http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Peter Howard
URSYS
13 Burwood Rd,
Burwood, NSW 2134

Ph: 02 8745 2816    Fax: 02 8745 2828



More information about the asterisk-users mailing list