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

Tue Jul 14 16:12:02 EST 2020

I see, Mark. It is true, in my case, I splitted webrtc to other
opensips (newer version) as our platform was too old.
I still think path module function should help:
https://opensips.org/docs/modules/3.1.x/path.html#func_add_path_received

Good luck

On Tue, Jul 14, 2020 at 11:48 AM Mark Allen <mark at allenclan.co.uk> wrote:
>
> 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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