On Friday 14 February 2014 16:31:20 Rob Moore wrote: > Anyway, I've had a look at the change you've suggested, unfortunately it > seems to have made little difference and the RTP proxy is still trying to > send traffic to the client SIPPhones private ip address and not the > firewall. > > It's probably worth clarifying (seen as the diagram got mangled) that I > don't have an RTPproxy at our client site only in our data centre paid > with our kamailio.
Well, we have the same setup: U 109.235.34.226:42259 -> 109.235.32.42:5060 INVITE sip:0880100...@pisco.pocos.nl SIP/2.0. Via: SIP/2.0/UDP 10.0.3.175:5062;branch=z9hG4bK-54389de. From: "tryba" <sip:tr...@pisco.pocos.nl>;tag=ea2052858ec448f9o2. To: <sip:0880100...@pisco.pocos.nl>. Call-ID: 20654a75-2dca31c9@10.0.3.175. CSeq: 101 INVITE. Max-Forwards: 70. Contact: "tryba" <sip:tryba@10.0.3.175:5062>. Expires: 240. User-Agent: Linksys/SPA962-6.1.3(a). Content-Length: 205. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: replaces. Content-Type: application/sdp. . v=0. o=- 872251 872251 IN IP4 10.0.3.175. s=-. c=IN IP4 10.0.3.175. t=0 0. m=audio 16404 RTP/AVP 2 101. a=rtpmap:2 G726-32/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. ... U 109.235.32.42:5060 -> 109.235.34.226:42259 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.0.3.175:5062;rport=42259;received=109.235.34.226;branch=z9hG4bK-54389de. Record-Route: <sip:109.235.32.42;lr=on;ftag=ea2052858ec448f9o2;vsf=AAAAAF9BSFZRc0RZQ1NfHjAfChwQQUAcb2Nvcy5ubA--;nat=yes;vsf=Q2lwOnRyeWJhQHBpc2NvLnBvY29zLm5s;nat=yes>. From: "tryba" <sip:tr...@pisco.pocos.nl>;tag=ea2052858ec448f9o2. To: <sip:0880100...@pisco.pocos.nl>;tag=as15267783. Call-ID: 20654a75-2dca31c9@10.0.3.175. CSeq: 101 INVITE. Server: Pocos. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Contact: <sip:+31880100...@109.235.32.xx>. Content-Type: application/sdp. Content-Length: 211. . v=0. o=root 87313901 87313901 IN IP4 109.235.32.42. s=Asterisk PBX 1.6.2.9-2+squeeze11. c=IN IP4 109.235.32.42. t=0 0. m=audio 45940 RTP/AVP 2. a=rtpmap:2 G726-32/8000. a=ptime:20. a=sendrecv. a=nortpproxy:yes. The natted client will start the RTP stream towards 109.235.32.42:45940 and after about 5-10 packets kamailio/rtpproxy will start sending to the source of the incoming steam. Attached a flow from an other test call. As far as I know the only change I made was moving the rtpproxy_manage() a bit. Maybe I'll have some time monday to generate a small proof of concept config, if nobody gives the answer before that time. -- POCOS B.V. - Croy 9c - 5653 LC Eindhoven Telefoon: 040 293 8661 - Fax: 040 293 8658 http://www.pocos.nl/ - http://www.sipo.nl/ K.v.K. Eindhoven 17097024
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