[asterisk-users] Wierd RTP issue

[Joshua Colp]

Richard Kenner wrote:
>> Not that many RTP packets are required. It's just important to see the
>> SIP signaling and where traffic is coming/going from with the network
>> topology in mind. That way a clearer picture of where it's saying media
>> should go to, where it's sending media from, etc can be gleamed. Once
>> that is figured out then the problem can be isolated.
>
> OK, I reproduced it on this machine.  It's a total of only 1293
> packets, taken on this end.  First call didn't work: I heard nothing
> coming inbound.  Second call worked, well enough that there was feedback
> (both phones and the desktop were in the same room).

Few suggestions:

1. Remove allow=gsm from your sip.conf and reload
2. Disable ZRTP in Jitsi by going into Options -> Accounts -> Selecting 
account -> Edit -> Security -> Uncheck "Enable support to encrypt calls".

See if that improves the situation.

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org