[sr-dev] No Media in SIP Incoming calls

Prashant Gupta prashant at farmguide.in
Wed Jan 9 08:14:57 CET 2019

I have the following architecture - SIP provider <-> Kamailio <-> Asterisk
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:")
this in my kamailio cfg but I don;t know which port to use here.
Any suggestions?
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