[SR-Users] rtpproxy sdp

[Daniel-Constantin Mierla]

Hello,

is Asterisk on a private IP and RTPEngine has to do bridging between
public and private networks?

Getting ngrep or pcap with sip traffic on kamailio server for a call
that doesn't work can help us figure out if something is not done for a
proper rtp relaying.

Cheers,
Daniel


On 15.09.17 05:44, Isravel Raja Thangamani wrote:
> Hi 
>
>  
> Thanks in advance if anyone can point me in the correct direction .
>
> I have kamailio running in front of some asterisk.  SIP endpoint
> register to their asterisk PBX via Kamailio dispatcher module. I'm running
> rtpengine with a Wan and private interface to bridge audio stream between
> these endpoints on the WAN to asterisk PBX running on LAN IP behind
> Kamailio.
>
> Calls from ext to ext work fine audio both directions , calls outbound to
> PSTN via SIP trunk to SIP provider via trunk on asterisk work fine audio
> both directions. But incoming calls via SIP provider  no audio from
> external caller to the asterisk ext neither asterisk to external caller
>
> I reckon I have something wrong in my Kamailio.cfg . if I register an ext
> direct to asterisk I get audio both ways on incoming calls. And rtp logs
>
> I think my mistake in somewhere in the cfg below. 
>
> Do I need to handle invites from the backend asterisk servers different that
> invites from sip endpoints?
>
>
>
>
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-- 
Daniel-Constantin Mierla
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