Hi Daniel,
yes, I hear the audio going both ways ok. In the meantime I succeeded to record something by rtpproxy but in best case only one direction, I can hear just the callee, but not the caller. In the meantime I have used start_recording in route[NATMANAGE] but before rtpproxy_manage("co"): -------------------------------------------------- if (is_method("INVITE") and (status=="200")) { start_recording(); } rtpproxy_manage("co"); -------------------------------------------------- After reading of your message I tested also with start_recording after rtpproxy_manage("co"): -------------------------------------------------- rtpproxy_manage("co"); if (is_method("INVITE") and (status=="200")) { start_recording(); } -------------------------------------------------- But the result was the same: I have 2 SIP users: 31 (X-Lite on Windows, static address 192.168.0.11) 35 (Android Zoiper, IP address 192.168.0.29 (by DHCP)) Kamailio has static address 192.168.0.13. When 35 calls 31 then I hear 1 direction in the recorded rtp file, the voice of callee. When 31 calls 35 then I don't hear anything in the recorded file. I don't understand why this difference when user 31 or 35 initiates a call!? But I hear audio conversation in both directions, just from the recorded file in one direction, or even nothing. I get 2 files per one call: <call-id>.rtp and <call-id>.rtcp How many files should rtpproxy produce for one call? Should also a file or two for 2nd channel be generated? How can I record both audio ways? Thank you Marko The whole function route[NATMANAGE]: # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if (is_method("INVITE") and (status=="200")) { start_recording(); } rtpproxy_manage("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { set_contact_alias(); } } #!endif return; } ----- Original Message ----- From: Daniel-Constantin Mierla <mico...@gmail.com> To: Marko Tirs <marko.t...@yahoo.com>; Kamailio (SER) - Users Mailing List <sr-users@lists.sip-router.org> Sent: Tuesday, March 14, 2017 5:23 PM Subject: Re: [SR-Users] rtpproxy doesn't record after start_recording Hello, I think you can call the start recording function just after rtpproxy_manage(). If you don't use the start recording function, is the audio going both ways without problems? Cheers, Daniel _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users