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

[Mark Allen]

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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