[OpenSIPS-Users] SIP Chronicles #7, Featuring Ken Rice

[Maxim Sobolev]

Hey, OpenSource folks this is just a friendly reminder. If you liked any of
our first 6 episodes of SIP Chronicles, please consider putting some time
aside this Saturday at 4:30 UTC to see Ken Rice talking about FreeSWITCH,
OpenSIPS and other projects that he is working on.

https://youtu.be/176gzKQOoSQ

See you soon!

-Max
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Re: [OpenSIPS-Users] rtpengine aborting loop when reinvite is sent

[Mario San Vicente]

Hello Alex,

I tried the flag loop-protect and it works.

I get the following message on the logs:

rtpengine[22968]: INFO: [fd72c95927567a5b]: Ignoring message as SDP has
already been processed by us

Thanks for your inputs!

Regards


On Fri, Jul 17, 2020 at 4:15 PM Alex Balashov 
wrote:

> I think one angle on the problem is to identify why RTPEngine is being
> invoked twice for the same SDP (most likely the same SIP message).
> That's best done by looking at the content of the reinvite itself. Focus
> especially on the Request URI and the Route header.
>
> I am not sure if the RTPEngine control module in OpenSIPS supports the
> loop-protect option, or whether specific module support for any command
> is required in order to use it.
>
> -- Alex
>
> On 7/17/20 5:06 PM, Mario San Vicente wrote:
> > Thanks for your explanation Alex,
> >
> > Actually i compiled the latest..git clone
> > https://github.com/sipwise/rtpengine.git
> >
> > But still i dont have a clue on how to solve it, any advise?
> >
> > Thank you
> > Mario
> >
> > On Fri, Jul 17, 2020 at 2:34 PM Alex Balashov  > > wrote:
> >
> > This happens when an SDP body that has already been passed to
> > RTPEngine, and already adulterated by RTPEngine, is passed to it yet
> > again.
> >
> > Newer versions of RTPEngine have a loop protection feature to deal
> > with it. It involves injecting an unregistered a=rtpengine attribute
> > into the SDP, to say “I was already here”.
> >
> > Most common cause on in-dialog requests (e.g. reinvites) is that
> > next hop is set to proxy itself due to RFC 2543 endpoint behaviour.
> > So, the proxy forwards the reinvite to itself and, unsurprisingly,
> > invokes RTPEngine again. Many times this can be fixed on the
> endpoint.
> >
> > —
> > Sent from mobile, with due apologies for brevity and errors.
> >
> >> On Jul 17, 2020, at 3:19 PM, Mario San Vicente
> >> mailto:mrsanvice...@gmail.com>> wrote:
> >>
> >> 
> >> Hello Everyone,
> >>
> >> I have a scenario, where opensips is working with rtpengine . Call
> >> connects fine with audio.
> >>
> >> I have enabled reinvite pinging to keep the call alive ; when the
> >> reinvite is sent to both legs of the calls i get  error messages
> >> and NO AUDIO.  The reinvite has full SPD body.
> >>
> >> some config related:
> >>
> >> modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2123
> >> ")
> >> modparam("dialog", "reinvite_ping_interval", 60)
> >> record_route();
> >>  create_dialog("rR");
> >>
> >> intermitent LOGS, until server is reboot:
> >>
> >>  rtpengine[21823]: ERR: Too many packets in UDP receive queue
> >> (more than 50), aborting loop. Dropped packets possible
> >>  rtpengine[21823]: WARNING: More than 30 duplicate packets
> >> detected, dropping packet to avoid potential loop
> >>
> >> Any idea what might be wrong?
> >>
> >> thanks you in advance!
> >>
> >> --
> >> Mario San Vicente
> >> Cheers!
> >> ___
> >> 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
> >
> >
> >
> > --
> > Mario San Vicente
> > Cheers!
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
> --
> Alex Balashov | Principal | Evariste Systems LLC
>
> Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>


-- 
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Cheers!
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Re: [OpenSIPS-Users] rtpengine aborting loop when reinvite is sent

[Alex Balashov]

I think one angle on the problem is to identify why RTPEngine is being 
invoked twice for the same SDP (most likely the same SIP message). 
That's best done by looking at the content of the reinvite itself. Focus 
especially on the Request URI and the Route header.


I am not sure if the RTPEngine control module in OpenSIPS supports the 
loop-protect option, or whether specific module support for any command 
is required in order to use it.


-- Alex

On 7/17/20 5:06 PM, Mario San Vicente wrote:

Thanks for your explanation Alex,

Actually i compiled the latest..git clone 
https://github.com/sipwise/rtpengine.git


But still i dont have a clue on how to solve it, any advise?

Thank you
Mario

On Fri, Jul 17, 2020 at 2:34 PM Alex Balashov > wrote:


This happens when an SDP body that has already been passed to
RTPEngine, and already adulterated by RTPEngine, is passed to it yet
again.

Newer versions of RTPEngine have a loop protection feature to deal
with it. It involves injecting an unregistered a=rtpengine attribute
into the SDP, to say “I was already here”.

Most common cause on in-dialog requests (e.g. reinvites) is that
next hop is set to proxy itself due to RFC 2543 endpoint behaviour.
So, the proxy forwards the reinvite to itself and, unsurprisingly,
invokes RTPEngine again. Many times this can be fixed on the endpoint.

—
Sent from mobile, with due apologies for brevity and errors.


On Jul 17, 2020, at 3:19 PM, Mario San Vicente
mailto:mrsanvice...@gmail.com>> wrote:


Hello Everyone,

I have a scenario, where opensips is working with rtpengine . Call
connects fine with audio.

I have enabled reinvite pinging to keep the call alive ; when the
reinvite is sent to both legs of the calls i get  error messages 
and NO AUDIO.  The reinvite has full SPD body.


some config related:

modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2123
")
modparam("dialog", "reinvite_ping_interval", 60)
record_route();
 create_dialog("rR");

intermitent LOGS, until server is reboot:

 rtpengine[21823]: ERR: Too many packets in UDP receive queue
(more than 50), aborting loop. Dropped packets possible
 rtpengine[21823]: WARNING: More than 30 duplicate packets
detected, dropping packet to avoid potential loop

Any idea what might be wrong?

thanks you in advance!

-- 
Mario San Vicente

Cheers!
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--
Mario San Vicente
Cheers!

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--
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Re: [OpenSIPS-Users] rtpengine aborting loop when reinvite is sent

[Mario San Vicente]

Thanks for your explanation Alex,

Actually i compiled the latest..git clone
https://github.com/sipwise/rtpengine.git

But still i dont have a clue on how to solve it, any advise?

Thank you
Mario

On Fri, Jul 17, 2020 at 2:34 PM Alex Balashov 
wrote:

> This happens when an SDP body that has already been passed to RTPEngine,
> and already adulterated by RTPEngine, is passed to it yet again.
>
> Newer versions of RTPEngine have a loop protection feature to deal with
> it. It involves injecting an unregistered a=rtpengine attribute into the
> SDP, to say “I was already here”.
>
> Most common cause on in-dialog requests (e.g. reinvites) is that next hop
> is set to proxy itself due to RFC 2543 endpoint behaviour. So, the proxy
> forwards the reinvite to itself and, unsurprisingly, invokes RTPEngine
> again. Many times this can be fixed on the endpoint.
>
> —
> Sent from mobile, with due apologies for brevity and errors.
>
> On Jul 17, 2020, at 3:19 PM, Mario San Vicente 
> wrote:
>
> 
> Hello Everyone,
>
> I have a scenario, where opensips is working with rtpengine . Call
> connects fine with audio.
>
> I have enabled reinvite pinging to keep the call alive ; when the reinvite
> is sent to both legs of the calls i get  error messages  and NO AUDIO.  The
> reinvite has full SPD body.
>
> some config related:
>
> modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2123")
> modparam("dialog", "reinvite_ping_interval", 60)
> record_route();
>  create_dialog("rR");
>
> intermitent LOGS, until server is reboot:
>
>  rtpengine[21823]: ERR: Too many packets in UDP receive queue (more than
> 50), aborting loop. Dropped packets possible
>  rtpengine[21823]: WARNING: More than 30 duplicate packets detected,
> dropping packet to avoid potential loop
>
> Any idea what might be wrong?
>
> thanks you in advance!
>
> --
> Mario San Vicente
> Cheers!
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>


-- 
Mario San Vicente
Cheers!
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Re: [OpenSIPS-Users] rtpengine aborting loop when reinvite is sent

[Alex Balashov]

This happens when an SDP body that has already been passed to RTPEngine, and 
already adulterated by RTPEngine, is passed to it yet again.

Newer versions of RTPEngine have a loop protection feature to deal with it. It 
involves injecting an unregistered a=rtpengine attribute into the SDP, to say 
“I was already here”.

Most common cause on in-dialog requests (e.g. reinvites) is that next hop is 
set to proxy itself due to RFC 2543 endpoint behaviour. So, the proxy forwards 
the reinvite to itself and, unsurprisingly, invokes RTPEngine again. Many times 
this can be fixed on the endpoint.

—
Sent from mobile, with due apologies for brevity and errors.

> On Jul 17, 2020, at 3:19 PM, Mario San Vicente  wrote:
> 
> 
> Hello Everyone,
> 
> I have a scenario, where opensips is working with rtpengine . Call connects 
> fine with audio.
> 
> I have enabled reinvite pinging to keep the call alive ; when the reinvite is 
> sent to both legs of the calls i get  error messages  and NO AUDIO.  The 
> reinvite has full SPD body.
> 
> some config related:
> 
> modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2123")
> modparam("dialog", "reinvite_ping_interval", 60)
> record_route();
>  create_dialog("rR");
> 
> intermitent LOGS, until server is reboot:
> 
>  rtpengine[21823]: ERR: Too many packets in UDP receive queue (more than 50), 
> aborting loop. Dropped packets possible
>  rtpengine[21823]: WARNING: More than 30 duplicate packets detected, dropping 
> packet to avoid potential loop
> 
> Any idea what might be wrong?  
> 
> thanks you in advance!
> 
> -- 
> Mario San Vicente
> Cheers!
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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[OpenSIPS-Users] rtpengine aborting loop when reinvite is sent

[Mario San Vicente]

Hello Everyone,

I have a scenario, where opensips is working with rtpengine . Call connects
fine with audio.

I have enabled reinvite pinging to keep the call alive ; when the reinvite
is sent to both legs of the calls i get  error messages  and NO AUDIO.  The
reinvite has full SPD body.

some config related:

modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2123")
modparam("dialog", "reinvite_ping_interval", 60)
record_route();
 create_dialog("rR");

intermitent LOGS, until server is reboot:

 rtpengine[21823]: ERR: Too many packets in UDP receive queue (more than
50), aborting loop. Dropped packets possible
 rtpengine[21823]: WARNING: More than 30 duplicate packets detected,
dropping packet to avoid potential loop

Any idea what might be wrong?

thanks you in advance!

-- 
Mario San Vicente
Cheers!
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[OpenSIPS-Users] siptrace - dual HEP destinations

[solarmon]

Hi,

I was able to get siptrace sent to a single HEP server. Now I would like to
be able to send to a secondary server as an additional destination.

I thought I could just create additional hep_id and trace_id entries and
run sip_trace() twice, but this didn't work and only the second HEP server
received traffic.

The config I used for proto_hep was:

loadmodule "proto_hep.so"
modparam("proto_hep", "hep_id", "[homer]:9060;transport=tcp;version=3;")
modparam("proto_hep", "hep_id", "[homer2]:9060;transport=tcp;version=3;")

The config I used for siptrace was:

loadmodule "siptrace.so"
modparam("siptrace", "trace_on", 1)
modparam("siptrace", "trace_id", "[traceid]uri=hep:homer")
modparam("siptrace", "trace_id", "[traceid2]uri=hep:homer2")

And the sip_trace() call used were:

sip_trace("traceid","T","sip"); # stateful mode - in/out transaction
sip_trace("traceid2","T","sip"); # stateful mode - in/out
transaction

It is possible/probable that I have a misundertanding of how these
variables/tokens in the brackets work.

Please can somebody provide, or point me to, an example of how this
multiple HEP server destinations could be achieved, assuming it can be done


Thank you.
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[OpenSIPS-Users] Dynamic Routing Stats

[Mark Farmer]

Hi everyone

I have been asked to look into getting Nagios to monitor active calls on
specific Dynamic Routing gateways.

Looking at the drouting() docs I don't see any exported statistics.
Is there a way to do this please?

Many thanks
Mark.
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Re: [OpenSIPS-Users] Documentation error?

[Mark Allen]

Hi Răzvan - thanks for that info.

>From memory, I believe that I tried using the sample script and got errors
so then went to see if I could find a version of the tutorial for 3.0.

As for updating the wiki, I'm still finding my way around OpenSIPS and
haven't got it working properly yet. However, once I'm up to speed, if I
get a chance I'll suggest an update.

Cheers,

Mark

On Fri, 17 Jul 2020 at 12:21, Răzvan Crainea  wrote:

> Hi, Mark!
>
> Actually that's just a dangling page, generated by our scripts for
> "creating" a new version. Did you get to that page through a link or
> something? As the Index page[1] doesn't reference it.
> TBH we weren't planning of updating the tutorial, since nothing major
> has changed, there are only some syntactic changes (according to my
> analysis). I do agree an updated version would be better, but due to
> lack of resources, this hasn't reached our priority list.
> Nevertheless, that is a wiki page, and any contribution is more than
> welcome.
>
> [1] https://www.opensips.org/Documentation/Tutorials#toc9
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 7/16/20 4:03 PM, Mark Allen wrote:
> > Not sure where to report this, so apologies if it's in the wrong place.
> >
> > The tutorial for Web Sockets with 3.0 looks to be wrong when running
> > 3.0.2. Example script is full of obsolete commands, modules and
> > variables - not very helpful. Will this be rectified on release of 3.1
> > Stable?
> >
> > https://opensips.org/Documentation/Tutorials-WebSocket-3-0
> >
> >
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
> ___
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>
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Re: [OpenSIPS-Users] negative ACK

[Slava Bendersky via Users]

Hello Johan, 
That google cloud but I am suspect some sort of NAT problem. I can register and 
place call, but loose route causing some routing issue. 

volga629 

From: "johan"  
To: "volga629"  
Sent: Friday, July 17, 2020 4:21:47 AM 
Subject: Fwd: [OpenSIPS-Users] negative ACK 




Is there a reason why you don't use a normal listener ? 
And please run netstat -tulpn as I believe that port 8443 is closed. 

wkr, 
 Forwarded Message  Subject:[OpenSIPS-Users] negative ACK 
Date:   Thu, 16 Jul 2020 22:40:05 -0400 (EDT) 
From:   Slava Bendersky via Users [ mailto:users@lists.opensips.org | 
 ] 
Reply-To:   Slava Bendersky [ mailto:volga...@networklab.ca | 
 ] , OpenSIPS users mailling list [ 
mailto:users@lists.opensips.org |  ] 
To: OpenSIPS users mailling list [ mailto:users@lists.opensips.org | 
 ] 

Hello Everyone, 
What possible reason that port portion is ignored here in loose route with ACK 
method. 
That latest 3.1-dev 

Also I wonder if possible maintain aliases domains in database. 

I set an alias 

Aliases: 
wss: gk5ix.doctor.lan:8443 

Jul 16 22:37:07 [263361] NOTICE:core:main: config file ok, exiting... 

Jul 16 22:22:45 dev1-opensips /usr/sbin/opensips[263213]: 
DBG:core:parse_to_param: tag=cqbimfskge 
Jul 16 22:22:45 dev1-opensips /usr/sbin/opensips[263213]: DBG:core:_parse_to: 
end of header reached, state=29 
Jul 16 22:22:45 dev1-opensips /usr/sbin/opensips[263213]: DBG:core:_parse_to: 
display={}, ruri={ sip:64...@gk5ix.doctor.lan:8443} 
Jul 16 22:22:45 dev1-opensips /usr/sbin/opensips[263213]: 
DBG:core:get_hdr_field:  [50]; uri=[sip:64...@gk5ix.doctor.lan:8443] 
Jul 16 22:22:45 dev1-opensips /usr/sbin/opensips[263213]: 
DBG:core:get_hdr_field: to body [] 
Jul 16 22:22:45 dev1-opensips /usr/sbin/opensips[263213]: 
DBG:core:get_hdr_field: cseq : <6492>  
Jul 16 22:22:45 dev1-opensips /usr/sbin/opensips[263213]: 
DBG:core:get_hdr_field: content_length=0 
Jul 16 22:22:45 dev1-opensips /usr/sbin/opensips[263213]: 
DBG:core:get_hdr_field: found end of header 
Jul 16 22:22:45 dev1-opensips /usr/sbin/opensips[263213]: 
DBG:sipmsgops:has_totag: totag found 
Jul 16 22:22:45 dev1-opensips /usr/sbin/opensips[263213]: 
DBG:core:parse_headers: flags=200 
Jul 16 22:22:45 dev1-opensips /usr/sbin/opensips[263213]: DBG:rr:is_preloaded: 
No 
Jul 16 22:22:45 dev1-opensips /usr/sbin/opensips[263213]: 
DBG:core:grep_sock_info_ext: checking if host==us: 16==11 && [gk5ix.doctor.lan] 
== [10.142.0.15] 
Jul 16 22:22:45 dev1-opensips /usr/sbin/opensips[263213]: 
DBG:core:grep_sock_info_ext: checking if port 5060 matches port 5060 
Jul 16 22:22:45 dev1-opensips /usr/sbin/opensips[263213]: 
DBG:core:grep_sock_info_ext: checking if host==us: 16==11 && [gk5ix.doctor.lan] 
== [10.142.0.15] 
Jul 16 22:22:45 dev1-opensips /usr/sbin/opensips[263213]: 
DBG:core:grep_sock_info_ext: checking if port 5060 matches port 5060 
Jul 16 22:22:45 dev1-opensips /usr/sbin/opensips[263213]: 
WARNING:rr:after_strict: no socket found to match RR [1][gk5ix.doctor.lan:5060] 
Jul 16 22:22:45 dev1-opensips /usr/sbin/opensips[263213]: DBG:rr:after_strict: 
Next hop: 
'sip:104.196.55.209:8443;transport=wss;lr;ftag=vdi4gq47qa064.90d9c317' is loose 
rout 
er 


Any help thank you. 

volga629 

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Re: [OpenSIPS-Users] Documentation error?

[Răzvan Crainea]

Hi, Mark!

Actually that's just a dangling page, generated by our scripts for 
"creating" a new version. Did you get to that page through a link or 
something? As the Index page[1] doesn't reference it.
TBH we weren't planning of updating the tutorial, since nothing major 
has changed, there are only some syntactic changes (according to my 
analysis). I do agree an updated version would be better, but due to 
lack of resources, this hasn't reached our priority list.
Nevertheless, that is a wiki page, and any contribution is more than 
welcome.


[1] https://www.opensips.org/Documentation/Tutorials#toc9

Best regards,

Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 7/16/20 4:03 PM, Mark Allen wrote:

Not sure where to report this, so apologies if it's in the wrong place.

The tutorial for Web Sockets with 3.0 looks to be wrong when running 
3.0.2. Example script is full of obsolete commands, modules and 
variables - not very helpful. Will this be rectified on release of 3.1 
Stable?


https://opensips.org/Documentation/Tutorials-WebSocket-3-0



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[OpenSIPS-Users] Segfault bei TLS certificates in MySQL DB

[xaled]

Hi,



I got server and client tls config working in opensips.cfg and now trying to
get it running with mysql DB.



I tried to use the latest opensips-cp (I know that 3.1 is not officially
supported) to import client certificate data that I created using
opensips-cli and getting a segfault. I can see the imported certificate
using MySQL cli.



Maybe I should try importing client cert from cli? What would be the correct
way to import certificate in to OpenSIPS mysql table from linux shell?



9 02:29:30 test /usr/sbin/opensips[8551]: INFO:tls_mgm:init_tls_dom:
Processing TLS domain 'test-cli'

Jul  9 02:29:30 test /usr/sbin/opensips[8551]: DBG:tls_mgm:init_tls_dom: no
DH params file for tls domain 'test-cli' defined, using default '(null)'

Jul  9 02:29:30 test /usr/sbin/opensips[8551]: NOTICE:tls_mgm:init_tls_dom:
No EC curve defined

Jul  9 02:29:30 test /usr/sbin/opensips[8551]: DBG:tls_mgm:init_tls_dom:
cipher list null ... setting default

Jul  9 02:29:30 test /usr/sbin/opensips[8551]:
INFO:tls_mgm:get_ssl_ctx_verify_mode: server verification activated.

Jul  9 02:29:30 test /usr/sbin/opensips[8551]: NOTICE:tls_mgm:init_tls_dom:
no CA dir for tls 'test-cli' defined, using default '/etc/pki/CA/'

Jul  9 02:29:30 test /usr/sbin/opensips[8551]: NOTICE:tls_mgm:init_tls_dom:
no crl for tls, using none

Jul  9 02:29:30 test /usr/sbin/opensips[8551]: DBG:core:count_module_procs:
modules require 2 extra processes

Jul  9 02:29:30 test /usr/sbin/opensips[8551]: CRITICAL:core:sig_usr:
segfault in attendant (starter) process!



Thanks,

Xaled

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Re: [OpenSIPS-Users] ms teams outgoing calls fails

[Pasan Meemaduma via Users]

Hi Alexey,
Thanks for the prompt reply. your are a legend as always :). Sorry for not 
digging into old emails. The searches I did, didn't come with above posts.
Thank you again.
Regards,Pasan

   Distinguishing What && How ! 

On Friday, 17 July 2020, 11:27:16 am GMT+5:30, Alexey Vasilyev 
 wrote:  
 
 Hi Pasan,

This was explained here:
http://opensips-open-sip-server.1449251.n2.nabble.com/TLS-handshake-failure-td7619394.html




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

[Mark Allen]

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

[Alexey Vasilyev]

Hi Mark,

try this:

if (nat_uac_test("123")) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
fix_nated_contact();
}
}



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Alexey Vasilyev
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