Re: [SR-Users] Kamailio proxy at Amazon EC2 (behind NAT)

Hello. I'm trying to set up this (v 4.2 stable): peer <--> ec2 <--kamailio+rtpengine--> asterisk scheme I use advertised adress for SIP and WS connections. The problem is that on SIP I get one way audio - I can receive audio from asterisk, but I can't transmit audio there - my SIP UA tries to send data to Kamailio-s local EC2 IP.

In case of WebRTC I get lot's of erros: Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: <core> [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via header Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [forward.c:584]: forward_request(): building failed Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) The call reaches Asterisk, but not vice-versa. No media is being transferred. Rtpengine flags I use: For SIP: rtpengine_manage("trust-adress replace-origin replace-session-connection RTP/AVP"); For WS: rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force RTP/AVP"); Do you have any ideas how ti fix that? I also make REGFWD's to Asterisk -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users