[asterisk-dev] RTP interop with Sonus: hack

[Kristian Kielhofner]

On 1/7/09, Joshua Colp <jcolp at digium.com> wrote:
> ----- "Kristian Kielhofner" <kristian.kielhofner at gmail.com> wrote:
>
>  >
>  > Replying to myself...  I think I got it.  In getting all caught up on
>  > timestamps and sequence numbers I forgot to check the SSRC.  Sure
>  > enough, as of Josh's latest fixes RFC2833 events now have a different
>  > SSRC than the voice RTP packets.  Not sure if this was intended but
>  > that could very well be the "new" problem.
>  >
>  > I'll keep looking...
>
>
> I pulled up the wireshark traces you linked to and did not see this... the SSRC was the same as the following voice packets and everything indeed looks fine.
>
>
>  --
>  Joshua Colp
>  Digium, Inc. | Software Developer
>  445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
>  Check us out at:  www.digium.com  & www.asterisk.org
>

Me again...

I think Sonus is expecting the first RFC 2833 event (start) to have a
duration of 0 and increment from there.  Asterisk currently starts at
160 and increments (by 160).  Sonus starts at 0 and increments by 80.

I've confirmed that Asterisk is able to receive the events from Sonus
properly.  Sonus just doesn't like what Asterisk is sending...

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com