Problem with signalling - RTP gets lost! Rtpproxy not working properly?
I am absolutely stuck ... cause this happens in a live environement.
I have the following situation
A calls B over carrier 1 - number is not valid and I get back error 404 from carrier and now freeswitch should play a message saying: "number not valid". But from carrier 1 I get back an RTP stream that is useless [1] - and if the correct streams opens from freeswitch - this does not get back to A [2].
I tested with rtpproxy on Kamailio - and all the rtp streams arrive at the Kamailio - but they cannot be "connected" correctly.
I guess the problem is the 183 I get back from carrier 1 - after whitch rtp is opened. Or there is a wrong sdp singallisation if the "correct" stream arrives [3].
Sorry - I cannot get a solution - but I could provide various tcpdumps and pcaps.
A Kamailio Carrier 1 Freeswitch
INVITE ---------------------> 100 Your call is important <--------------------- INVITE ------------------------> 100 Trying <------------------------ 183 Session Progress SDP <------------------------ 183 Session Progress SDP <------------------------ RTP <================= RTP <================= RTP =================> RTP =================> [1]
404 not found <--------------------------- ACK ---------------------------->
INVITE ---------------------------------------------------------> 100 Trying <-------------------------------------------------------- 200 OK SDP <-------------------------------------------------------- [3] 200 OK SDP <--------------------------- RTP (Announcment - number not valid") [2] <=================================== ACK ----------------------------> ACK ----------------------------------------------------------> INFO ----------------------------> INFO ----------------------------------------------------------> 200 OK <-------------------------------------------------------- 200 OK <--------------------------- BYE ----------------------------> BYE ----------------------------------------------------------> 200 OK <-------------------------------------------------------- 200 OK <---------------------------
Hi,
this may make your config a little more tricky: You could instruct the RTPProxy/Mediaproxy-ng to "Trust" the IP in the SDP for the Originating call to PSTN and the Freeswitch (add "r" the flags). Have you verified/checked, that rtpproxy_manage() is really called for both the 183 and the 200 OK?
Thanks, Carsten
2013/11/28 Oliver Roth oliver.roth@triotel.ch:
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Carsten Bock -
Oliver Roth