[SR-Users] No Media in SIP Incoming calls

YAS0 CANER caner_yaso at hotmail.com
Wed Jan 9 08:48:43 CET 2019


1- Probably, there isnt any command to active rtpengine in your conf. check  your configuration that do you active rtpengine_manage/offer/answer() on INVITE
2- Rtpengine service is working?
3- Kamailio can connect to rtpengine service , you can see systemlog/message
4-  take a look your pcap , rtpengine can change  ip address  in sdp
5 - do you set "replace-origin replace-session-connection" parameters to rtpengine?

rtpengine doc : https://kamailio.org/docs/modules/devel/modules/rtpengine.html
rtpengine Module - kamailio.org<https://kamailio.org/docs/modules/devel/modules/rtpengine.html>
This is a module that enables media streams to be proxied via an RTP proxy. The only RTP proxy currently known to work with this module is the Sipwise rtpengine https ...

Good luck

From: sr-users <sr-users-bounces at lists.kamailio.org> on behalf of Prashant Gupta <prashant at farmguide.in>
Sent: Wednesday, January 9, 2019 10:36 AM
To: sr-users at lists.kamailio.org
Subject: [SR-Users] No Media in SIP Incoming calls

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:<>")
this in my kamailio cfg but I don;t know which port to use here.
Any suggestions?
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