Hello Mark, I had a similar challenge. Using "path" module on both opensips helps to overcome this problem. https://opensips.org/docs/modules/3.2.x/path.html
In your mid-registerer you need to enable path support. See "save" function params p0 and v. in your webrtc opensips use path module and function add_path_received On Tue, Jul 14, 2020 at 11:14 AM Mark Allen <m...@allenclan.co.uk> wrote: > > I'm new to OpenSIPS and I've hit a problem I can't find a way past > > We have a test setup with an OpenSIPS mid-registrar in front of an Asterisk > PBX. Mid-registrar is currently in mode 1 (registration throttling). We have > SIP and WebRTC endpoints that we want to use. > > Current state is: > > REGISTER: WebRTC webphone (Mizutech) -> OpenSIPS Mid-registrar -> Asterisk > = success > REGISTER: SIP softphone (LinPhone) -> OpenSIPS Mid-registrar -> Asterisk > = success > > INVITE: SIP softphone -> OpenSIPS -> Asterisk -> OpenSIPS -> SIP > softphone = success, call connects with audio both ways > INVITE: WebRTC webphone -> OpenSIPS -> Asterisk -> OpenSIPS -> SIP > softphone = success, call connects with audio both ways > INVITE: SIP softphone -> OpenSIPS -> Asterisk -> OpenSIPS -> WebRTC > webphone = fails with "476 Unresolvable destination" > > syslog messages: > ERROR:core:sip_resolvehost: forced proto 6 not matching sips uri > CRITICAL:core:mk_proxy: could not resolve hostname: "4xp44jxl0qq0.invalid" > ERROR:tm:uri2proxy: bad host name in URI > <sips:11001@4xp44jxl0qq0.invalid;rtcweb-breaker=yes;transport=wss> > ERROR:tm:t_forward_nonack: failure to add branches > > > Following past reports that I've found with a similar error, > fix_nated_contact() is run on INVITE messages just before rtpengine flags are > set and the t_relay() command, but it doesn't appear to make any difference. > If I change the t_relay() to t_relay(0x04,) to disable DNS Failover, I still > see the same errors in the log file. I've also checked the record in the > OpenSIPS DB "location" table and it seems to me that it has the correct > contact_id and contact info for the destination... > > contact_id: 2004383309156582802 > contact: sips:11001@4xp44jxl0qq0.invalid;rtcweb-breaker=yes;transport=wss > > I'm stuck on where I can go from here - any help very much appreciated! > > thx > > Mark > > > Setup: > OpenSIPS 3.0.2 on Debian Buster > RTPEngine Version: 8.4.0.0+0~mr8.4.0.0 > > INVITE: > 2020/07/14 14:22:06.176544 192.168.50.185:5060 -> 192.168.50.69:5060 > INVITE sip:11001@192.168.50.69:5060;ctid=2004383309156582802 SIP/2.0 > Via: SIP/2.0/UDP > 192.168.50.185:5060;rport;branch=z9hG4bKPj3e87a449-f4cc-4128-abbe-95706a1a44a0 > From: "11002" > <sip:11002@192.168.50.185>;tag=1c03916d-d086-479a-b984-ff5bbbf3aba8 > To: <sip:11001@192.168.50.69;ctid=2004383309156582802> > Contact: <sip:asterisk@192.168.50.185:5060> > Call-ID: d1524788-cac2-4bea-a905-4e17ba006688 > CSeq: 24456 INVITE > Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, > CANCEL, UPDATE, PRACK, MESSAGE, REFER > Supported: 100rel, timer, replaces, norefersub > Session-Expires: 1800 > Min-SE: 90 > P-Asserted-Identity: "11002" <sip:11002@192.168.50.185> > Max-Forwards: 70 > User-Agent: FPBX-15.0.16.63(16.9.0) > Content-Type: application/sdp > Content-Length: 411 > > v=0 > o=- 263255642 263255642 IN IP4 192.168.50.185 > s=Asterisk > c=IN IP4 192.168.50.185 > t=0 0 > m=audio 10292 RTP/AVPF 9 107 8 0 3 111 101 > a=rtpmap:9 G722/8000 > a=rtpmap:107 opus/48000/2 > a=fmtp:107 useinbandfec=1 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:111 G726-32/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=maxptime:20 > a=sendrecv > a=rtcp-mux > > > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
