[asterisk-users] canreinvite=yes and RTP dropping in and out

[Joshua Colp]

----- Original Message -----
From: Gary Richardson
[mailto:gary.richardson at gmail.com]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:asterisk-users at lists.digium.com]
Sent:
Wed, 02 Aug 2006 13:54:04 -0300
Subject: [asterisk-users] canreinvite=yes
and RTP dropping in and out


> Hey guys,
> 
> I'm having yet another strange problem. I've recently set canreinvite=yes,
> allowing the RTP streams to avoid our * server. Now, a few people are
> experience one way audio drops on internal calls. External calls are working
> fine (they re-invite directly to a Cisco router). Sometimes, if you wait 20
> seconds or more, the stream will resume. Flipping the person on and off hold
> won't resume the stream.
> 
> We're using 7960 phones. Enabled_vad is set to 0 (disabled). It doesn't seem
> to happen all of the time. There are no sip messages being exchanged when
> the stream stops or restarts.
> 
> Any suggestions?

If the audio is going directly there's not too much you can do to examine it. There may be software out there to sniff the data on your network and examine the RTP stream, maybe even see when it drops out (if it really does drop out, ie: stream actually stops). I know there's some Windows software out there capable of this as I picked a copy up while at Spring VON but you might need to look around. OH - can you also send a sip debug with the reinvites? I'm just curious to see the RTP information in the SDP.

> Thanks.

Joshua Colp
Digium