OK. On the Freeswitch side there is a double transport=tls but I can’t find out where that’s happening so I’ll remove it in the Opensips script
Here’s what Opensips sees from Freeswitch James Hogbin Director IP Sentinel t. +44 (0)20 3011 4150 m. +44 7786910895 w. https://www.ip-sentinel.com INVITE sip:08435577...@sbc.ip-sentinel.com:5091;transport=tls SIP/2.0 Via: SIP/2.0/TLS 13.80.245.144:5081;rport;branch=z9hG4bKK90SUgZ4ZmN1D Max-Forwards: 68 From: "James Hogbin" <sip:+442030114146@13.80.245.144>;tag=t4ce3rS0Fy3ga To: <sip:08435577...@sbc.ip-sentinel.com:5091> Call-ID: 86371d7b-0e76-1239-bdba-000d3aada04e CSeq: 20063606 INVITE Contact: <sip:gw+c6ff36e8-d3de-4fe0-9f1b-9da2888c43a9@13.80.245.144:5081;transport=tls;transport=tls;gw=c6ff36e8-d3de-4fe0-9f1b-9da2888c43a9> User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 1317 X-FS-Support: update_display,send_info Remote-Party-ID: "James Hogbin" <sip:+442030114146@13.80.245.144>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1589217436 1589217437 IN IP4 13.80.245.144 s=FreeSWITCH c=IN IP4 13.80.245.144 t=0 0 m=audio 17360 RTP/SAVP 9 0 8 101 13 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=crypto:1 {removed] a=ptime:20 m=audio 17360 RTP/AVP 9 0 8 101 13 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 Here is my script. I’m not fixing FROM or TO. I am fixing the number, the Record_Route & the duplicated transport=tls in the CONTACT if (is_method("INVITE") && !has_totag() && check_source_address(0)) { xlog("[ROUTE]Incoming call to MS: RURI=$ruri, SI=$si, M=$rm\n"); trace("tid"); strip(1); prefix("+44"); $rd="sip.pstnhub.microsoft.com<http://sip.pstnhub.microsoft.com>"; $rp=5061; $var(reg_input) = $ct; $var(reg)= "/transport=tls;transport=tls;/transport=tls;/g"; xlog("[WARNING]Applying reg exp $var(reg) to $var(reg_input)"); $var(reg_output) = $(var(reg_input){re.subst,$var(reg)}); remove_hf("Contact"); append_hf("Contact: $var(reg_output)"); record_route_preset("sbc.ip-sentinel.com:5091<http://sbc.ip-sentinel.com:5091>;transport=tls"); route(relay); The RTP_Proxy seems to be updating the SDP correctly And here is the invite to teams INVITE sip:+448435577...@sip.pstnhub.microsoft.com:5061;transport=tls SIP/2.0 Record-Route: <sip:sbc.ip-sentinel.com:5091;transport=tls;ftag=t4ce3rS0Fy3ga;lr;vsf=TxoZShEfAAYCAwMBAAUFBQJ2cUBMWx5HQhlGS19AXW5lbC5jb206NTA5MT4-;vst=TxoZSgoTAAcNAQQCAgUGd0FTI10AAAMDFhoaARsHQkMKDB9VRl9fRS5jb20+> Via: SIP/2.0/TLS 137.117.136.143:5091;branch=z9hG4bK9915.ccfe3523.0;i=ae435223 Via: SIP/2.0/TLS 13.80.245.144:5081;received=13.80.245.144;rport=36423;branch=z9hG4bKK90SUgZ4ZmN1D Max-Forwards: 68 From: "James Hogbin" <sip:+442030114...@sbc.ip-sentinel.com:5091>;tag=t4ce3rS0Fy3ga To: <sip:+448435577...@sip.pstnhub.microsoft.com> Call-ID: 86371d7b-0e76-1239-bdba-000d3aada04e CSeq: 20063606 INVITE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 1339 X-FS-Support: update_display,send_info Remote-Party-ID: "James Hogbin" <sip:+442030114146@13.80.245.144>;party=calling;screen=yes;privacy=off Contact: <sip:gw+c6ff36e8-d3de-4fe0-9f1b-9da2888c43a9@13.80.245.144:5081;transport=tls;gw=c6ff36e8-d3de-4fe0-9f1b-9da2888c43a9> v=0 o=FreeSWITCH 1589217436 1589217437 IN IP4 137.117.136.143 s=FreeSWITCH c=IN IP4 137.117.136.143 t=0 0 m=audio 10124 RTP/SAVP 9 0 8 101 13 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=crypto:1 {removed} a=ptime:20 m=audio 19352 RTP/AVP 9 0 8 101 13 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 a=nortpproxy:yes I don’t even get a response from Teams now so I think I’m now going backwards even though I think I’ve implemented all the parts of the advice I’ve been given throughout this thread. 0.00 Freeswitch => INVITE => Opensips 0.01 Freeswitch <= 100 <= Opensips 0.01 Opensips => INVITE => MSTeams 30.0 Freeswitch <= 408 <= Opensips 30.0 Freeswitch => ACK => Opensips 31.4 Freeswitch => ACK => Opensips James IP Sentinel Disclaimer This communication is for the information of the person to whom it has been delivered and neither it nor any of its contents should be passed on to or used by any other person. IP Sentinel Ltd is a limited company registered in England and Wales under Registered Number 08648097. Registered Office: Newnhams Wood, Horsted Keynes, West Sussex, RH17 7BT. Disclaimer: Q3dhRSrm_disclaimer
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