Re: [SR-Users] No Media in SIP Incoming calls

Hello, 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.h… 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 ... kamailio.org Good luck Yasin CANER ________________________________ From: sr-users &lt;sr-users-bounces(a)lists.kamailio.org&gt; on behalf of Prashant Gupta &lt;prashant(a)farmguide.in&gt; Sent: Wednesday, January 9, 2019 10:36 AM To: sr-users(a)lists.kamailio.org Subject: [SR-Users] No Media in SIP Incoming calls 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?