[Asterisk-video] Asterisk & H.264
John Martin
John.Martin at AuPix.com
Mon May 29 02:27:34 MST 2006
Hi Marco,
It's a public holiday in the UK today so I can't do too much checking
at the moment. If you send me (email address below) a debug trace (type
"sip debug" on the asterisk CLI) or an Ethereal trace then I can have a
look at it for you. It may be that the Grandstream is expecting to see
the fmtp for H.264.
I would guess that if you're audio is bad then perhaps a lower quality
audio codec has been negociated (probably g.729). Try removing the G.729
capability from your peer definition to see if that helps.
You might also try increasing your maxcallbitrate to something higher
than 128, just to see if that makes any difference.
John
John Martin
John.Martin at AuPix.com
http://www.AuPix.com
-----Original Message-----
From: asterisk-video-bounces at lists.digium.com
[mailto:asterisk-video-bounces at lists.digium.com] On Behalf Of Marco
Ciacci
Sent: 29 May 2006 07:46
To: Discussion of video media support in Asterisk
Subject: RE: [Asterisk-video] Asterisk & H.264
At 17.45 26/05/2006, you wrote:
>Hi Marco,
> Trunk now supports videosupport and maxcallbitrate on a per peer
>basis.
>It wouldn't hurt setting maxcallbitrate too.
>
>So you'll need something like:
>[general]
>videosuport=yes
>maxcallbitrate=512
>
>and maybe
>
>[extension]
>videosupport=yes
>maxcallbitrate=128
I've wrote out all in my files, but doesn't work! :(
My extension for example has:
[210]
username=210
type=friend
secret=xxxxxxxx
qualify=no
port=5060
nat=never
host=dynamic
dtmfmode=rfc2833
context=from-internal
callerid=device <210>
canreinvite=yes
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=h264
allow=h261
allow=h263
allow=h26p
videosupport=yes
maxcallbitrate=128
With allow=h261, h263 and h263p the audio is also bad, with only h264
the audio is normal but in any case without video :-(
What could be wrong?
--
Marco Ciacci
Linux & Networking - Silog Srl. (SI)
Tel. 0577271840 - mciacci at silog.it
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