[OpenSIPS-Users] SIP to WebRTC via OpenSIPS mid-registrar fails: forced proto 6 not matching sips uri

[Mark Allen]

Thanks Stas - I'll have a look at that.

For clarification, we only have one OpenSIPS server acting as
mid-registrar. Endpoints register through it to extensions on Asterisk, and
Asterisk acts as B2BUA for calls from one extension to another. We've got a
lot of additional functionality linked to the Asterisk server so our main
need for OpenSIPS is to reduce unnecessary load (e.g. re-REGISTER from
mobile devices).

On Tue, 14 Jul 2020 at 16:23, Stas Kobzar <staskobzar at gmail.com> wrote:

> 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 <mark at 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 at 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 at 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 at 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 at 192.168.50.185
> >;tag=1c03916d-d086-479a-b984-ff5bbbf3aba8
> > To: <sip:11001 at 192.168.50.69;ctid=2004383309156582802>
> > Contact: <sip:asterisk at 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 at 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
> >
> >
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>
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