Problem found Re: [asterisk-users] Headaches with Video over SIP

[Peter Howard]

On Tue, 2006-11-14 at 09:28 +1100, Peter Howard wrote:
> 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 got a log from a call under asterisk 1.4.0-beta3 attached. The
> behaviour was the same; the call connected and audio worked, but no
> video.
> 

Found the problem.  Some other people on the list might be interested in
the problem.

Looking at the initial INVITE call from station 1 to asterisk and from
asterisk to station 2, you see the following discrepancy:

>From Polycom to asterisk:
a=rtpmap:109 H264/90000
a=rtpmap:96 H263-1998/90000

>From asterisk to other Polycom:
a=rtpmap:103 h263-1998/90000
a=rtpmap:99 H264/90000

The latter numbers match the numbers in rtp.c, which clearly don't agree
with the Polycom's opinion on life.  However the other Polycom seems to
accept asterisk's version, while the first Polycom is still working on
it's version.

I modified the initialisation of static_RTP_PT in rtp.c to match the
Polycom values and, hey presto!  I have video working between the two
stations.

However, this doesn't look like a "proper" solution, if there are
multiple opinions out there of what h263p and h264 (at least) should be
mapped to.  There's also the array current_RTP_PT in rtp.c  What is that
used for?  I would have thought that either:
1) Asterisk should tell _both_ ends what mapping to use,
2) Asterisk should update it's own mapping based on what it's told
(though this would have to be on a call-by-call basis)



> 
> 
> > 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
> >         
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-- 
Peter Howard
URSYS
13 Burwood Rd,
Burwood, NSW 2134

Ph: 02 8745 2816    Fax: 02 8745 2828