Re: [OpenSIPS-Users] SIP to WebRTC via OpenSIPS mid-registrar fails: forced proto 6 not matching sips uri
Hi Mark, is working for us more or less, because is strange situtation, if the extension WSS received a incoming call working good but if a put the call in on-hold inmediatly hangout, but if the same extension do the call to a external number or a SIP extension or another Webrtc extension work very good hold and unhold very good the call, i dont know how to identify what is the problem, we are trying to do this in Opensips because actually our Asterisk PBX receiving directly WSS conections some times have Flooding Error (From Traffic Valid). -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP to WebRTC via OpenSIPS mid-registrar fails: forced proto 6 not matching sips uri
Hi Juan Carlos - I feel your pain!
I've finished for the year so I don't have access to the code at the
moment. I'll have a look at this in the new year to see if I can post
something sensible! :-)
Short version is that it turned out to be a horrible challenge to get it
working as desired. My memory is that the difficulty in finding a suitable
workaround arose because of how OpenSIPS works (LUMPS system) when you make
changes in that they are only committed at the end. This means that if you
change $du to something new but if you then immediately xlog("$du") you'll
still see the original value - the change is only committed when you've
finished working on that SIP message.
In the WSS and mid-registrar case, it means that to workaround the wss
problem by using the path setting I had to loopback the SIP message to
OpenSIPS to then be able to see the Path value. I had to fudge some things
with code but it's working for us now. Maybe somebody else knows a more
straightforward way?
On Wed, 23 Dec 2020 at 15:16, juancarlosg6 wrote:
> Hi, iam having the same situation, can you copy the opensips.cfg as
> reference, iam having issue with incoming calls to webrtc extensions, my
> opensips version is 3.1 and is working with mid_registar, i can do calls to
> SIP extensions, SIP Trunk, but not working when incoming calls to extension
> Webrtc is loged throung Opensips using mid_registar, thank you.
>
>
>
> --
> Sent from:
> http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html
>
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>
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Re: [OpenSIPS-Users] SIP to WebRTC via OpenSIPS mid-registrar fails: forced proto 6 not matching sips uri
Hi, iam having the same situation, can you copy the opensips.cfg as reference, iam having issue with incoming calls to webrtc extensions, my opensips version is 3.1 and is working with mid_registar, i can do calls to SIP extensions, SIP Trunk, but not working when incoming calls to extension Webrtc is loged throung Opensips using mid_registar, thank you. -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP to WebRTC via OpenSIPS mid-registrar fails: forced proto 6 not matching sips uri
Hi Mark, can you share the configuration? i need to do something similar but without success, thank you. Using your example: REGISTER: SIP softphone (LinPhone) -> OpenSIPS Mid-registrar -> Asterisk = success REGISTER: WebRTC webphone (Mizutech) -> OpenSIPS Mid-registrar -> Asterisk = Fail -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP to WebRTC via OpenSIPS mid-registrar fails: forced proto 6 not matching sips uri
Hi Mark, Glad it helped. I actually did not know about chan_sip option supportpath because I was using additional opensips for wws to webrtc. Thanks for the details. Good luck On Thu, Jul 23, 2020 at 9:19 AM Mark Allen wrote: > > [SOLVED] > > Thanks Stas. Your workaround did solve the problem and I see that with 3.1 > path support is baked into mid-registrar module as options to > mid_registrar_save(). > > Once we added in the path module functionality, at first it didn't work. > Looking at sngrep traces we could see that the path information was appended > on the inbound route from OpenSIPS to Asterisk, but when Asterisk made the > call to the outbound destination it failed to include it as route info. This > was resolved by setting supportpath="yes" in sip.conf and worked with > CHAN_SIP. We tried to get it working with PJSIP without any joy but it's not > a priority for us at the moment so we'll have to investigate the cause later. > > Thanks for your help Stas, and thanks also to the others who took the time to > reply. > > cheers, > > Mark > > > On Tue, 14 Jul 2020 at 16:23, Stas Kobzar 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 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 >> > >> > 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" >> > ;tag=1c03916d-d086-479a-b984-ff5bbbf3aba8 >> > To: >> > Contact: >> > 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" >> > 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 >> > >> > >> >
Re: [OpenSIPS-Users] SIP to WebRTC via OpenSIPS mid-registrar fails: forced proto 6 not matching sips uri
[SOLVED] Thanks Stas. Your workaround did solve the problem and I see that with 3.1 path support is baked into mid-registrar module as options to mid_registrar_save(). Once we added in the path module functionality, at first it didn't work. Looking at sngrep traces we could see that the path information was appended on the inbound route from OpenSIPS to Asterisk, but when Asterisk made the call to the outbound destination it failed to include it as route info. This was resolved by setting supportpath="yes" in sip.conf and worked with CHAN_SIP. We tried to get it working with PJSIP without any joy but it's not a priority for us at the moment so we'll have to investigate the cause later. Thanks for your help Stas, and thanks also to the others who took the time to reply. cheers, Mark On Tue, 14 Jul 2020 at 16:23, Stas Kobzar 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 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 ;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" >;tag=1c03916d-d086-479a-b984-ff5bbbf3aba8 > > To: > > Contact: > > 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" > > 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 > ___ Users mailing list
Re: [OpenSIPS-Users] SIP to WebRTC via OpenSIPS mid-registrar fails: forced proto 6 not matching sips uri
Hi Alexey - thanks for responding.
I've seen past reports where NAT was causing this type of problem. I tried
your suggestion but, along with other tests such as forcing
fix_nated_register() or fix_nated_contact() on all messages, and after
trying Stas' suggestion, it still doesn't work for me. I believe that I've
followed the tutorials correctly but, as per another post of mine, there
are issues with the 3.0 tutorial scripts being out of date - so maybe the
problem lies somewhere in how I've tried to fix the script errors for 3.0.2?
It's very frustrating because I can look at the "location" table in MySQL
and see the information being correctly stored, it just appears that for
some reason when it looks up and retrieves the information it doesn't
recognise that the destination retrieved is a WebSocket and so tries to
treat it as a FQDN - or at least that's what it seems to be saying to me.
cheers,
Mark
On Fri, 17 Jul 2020 at 07:07, Alexey Vasilyev
wrote:
> Hi Mark,
>
> try this:
>
> if (nat_uac_test("123")) {
> if (is_method("REGISTER")) {
> fix_nated_register();
> } else {
> fix_nated_contact();
> }
> }
>
>
>
> -
> ---
> Alexey Vasilyev
> --
> Sent from:
> http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html
>
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>
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Re: [OpenSIPS-Users] SIP to WebRTC via OpenSIPS mid-registrar fails: forced proto 6 not matching sips uri
Hi Mark,
try this:
if (nat_uac_test("123")) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
fix_nated_contact();
}
}
-
---
Alexey Vasilyev
--
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Re: [OpenSIPS-Users] SIP to WebRTC via OpenSIPS mid-registrar fails: forced proto 6 not matching sips uri
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 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 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 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 >> > >> > 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" >> > ;tag=1c03916d-d086-479a-b984-ff5bbbf3aba8 >> > To: >> > Contact: >> > 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" >> > 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 > >
Re: [OpenSIPS-Users] SIP to WebRTC via OpenSIPS mid-registrar fails: forced proto 6 not matching sips uri
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 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 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 ;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" >;tag=1c03916d-d086-479a-b984-ff5bbbf3aba8 > > To: > > Contact: > > 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" > > 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 > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP to WebRTC via OpenSIPS mid-registrar fails: forced proto 6 not matching sips uri
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 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 > > 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" > ;tag=1c03916d-d086-479a-b984-ff5bbbf3aba8 > To: > Contact: > 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" > 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
