[sr-dev] No Media in SIP Incoming calls

[Daniel-Constantin Mierla]

Hello,

sr-dev mailing list is for discussions about writing code/developing
Kamailio. For question on how to use kamailio and how to configure it
for various scenarios, send your messages to
sr-users at lists.kamailio.org. That list has more members and the
developers are also there, so the chance to get better answer is higher.

Cheers,
Daniel

On 09.01.19 08:14, Prashant Gupta wrote:

> Hi,
> I have the following architecture - SIP provider <-> Kamailio <->
> Asterisk servers
> Currently I have everything setup and incoming calls from Sip are
> routed to my asterisk server. The issue is however that when I answer
> the call, there is no media in the call. I have tried connecting with
> a normal local extension(not SIP,eg 1001) and there is a normal flow
> of media.
> When i try to sniff my connection via Wireshark on the asterisk
> server, there is an outflow of RTP packets but the same RTP traffic
> does not appear on the Wireshark of my Kamailio server connection. 
> I am not sure if this is an RTP engine issue and how to resolve this.
> I have -
> modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:45038
> <http://127.0.0.1:45038>")
> this in my kamailio cfg but I don;t know which port to use here.
> Any suggestions?
>
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-- 
Daniel-Constantin Mierla -- www.asipto.com
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