Hello Razvan, /Are you sure your clients are properly pinged by OpenSIPS? Are you setting the sipping_flag[1] for your natted clients?/ No i haven't it. I've enabled now, tcpdumping i see the pinging with the OPTIONS but no changes...
/What about the nat_bflag flag[2]?/ Yes, i've it. Doing ngrep in the opensips server, it is clear that the "gap" is somewhere internally, if you check the difference time between the received INVITE and the sending INVITE is +/- 10 secs... *T 2014/04/20 11:08:37.829732* 31.27.52.41:45283 -> X.X:X:X:5060 [AP] INVITE sip:[email protected];transport=tcp SIP/2.0. Via: SIP/2.0/TCP 31.27.52.41:45283;branch=z9hG4bK.AaFavZIjO;rport. From: <sip:[email protected]>;tag=fAtsL4CLE. To: sip:[email protected]. CSeq: 21 INVITE. Call-ID: mJnHuOO85u. Max-Forwards: 70. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Content-Length: 472. User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4). Contact: <sip:[email protected]:45283;transport=tcp>;+sip.instance="<urn:uuid:9a0048a3-5041-43f1-b72e-fc9c44580d60>". Proxy-Authorization: Digest realm="sip-lab.xxx.yyy", nonce="53538eb30000016a5c48fc66aef7eede3ef6dd076fbb460e", username="uuu", uri="sip:[email protected];transport=tcp", response="18e2b8c8f05f9a989ab10af89ac5c51b". . v=0. o=uuu 3644 3041 IN IP4 31.27.52.41. s=Talk. c=IN IP4 31.27.52.41. b=AS:380. t=0 0. m=audio 7076 RTP/AVP 123 119 111 110 0 8 101. a=rtpmap:123 opus/48000. a=fmtp:123 useinbandfec=1; usedtx=1. a=rtpmap:119 SILK/16000. a=rtpmap:111 speex/16000. a=fmtp:111 vbr=on. a=rtpmap:110 speex/8000. a=fmtp:110 vbr=on. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11. m=video 9078 RTP/AVP 103 99. a=rtpmap:103 VP8/90000. a=rtpmap:99 MP4V-ES/90000. a=fmtp:99 profile-level-id=3. *T 2014/04/20 11:08:47.846158* X.X:X:X:5060 -> Z.Z.Z.Z:50805 [AP] INVITE sip:[email protected]:50805;transport=tcp SIP/2.0. Record-Route: <sip:X.X.X:X;transport=tcp;lr;did=28f.923f4063;nat=yes>. Via: SIP/2.0/TCP X.X.X.X:5060;branch=z9hG4bK204b.07657247.1;i=c. Via: SIP/2.0/TCP 31.27.52.41:45283;received=31.27.52.41;branch=z9hG4bK.AaFavZIjO;rport=45283. From: <sip:[email protected]>;tag=fAtsL4CLE. To: sip:[email protected]. CSeq: 21 INVITE. Call-ID: mJnHuOO85u. Max-Forwards: 69. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Content-Length: 498. User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4). Contact: <sip:[email protected]:45283;transport=tcp>;+sip.instance="<urn:uuid:9a0048a3-5041-43f1-b72e-fc9c44580d60>". -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/UDP-vs-TCP-TLS-tp7590719p7590846.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
