Re: [OpenSIPS-Users] opensips and multiple rtpengine instances

[Vitalii Aleksandrov]

It happened that rtpengine with multiple rtp workers sometimes receives 
RTP packets ordered sequentially but forwards them a bit reordered 
introducing out-of-order frames. With one rtp worker everything works 
fine. The idea was to start a couple of single threaded rtpengines and 
loadbalance from opensips. But it looks like not with the current 
opensips/rtpengine module.
I remember that somebody from smartvox  opened an issue on this (it 
was either Pete Kelly or John Quick).


Pete,John
do you recall ?

Op vr 26 apr. 2019 om 15:15 schreef Vitalii Aleksandrov 
mailto:vitalik.v...@gmail.com>>:


Hello opensips users,

has anybody tried to configure opensips with multiple rtpengine
sockets?

Logically thinking offer/answer from initial INVITE/OK and from all
in-dialog messages must go to the same instance and opensips can load
balance only initial INVITEs.
I briefly compared opensips/rtpengine and kamaio/rtpengine modules
and
found that this logic is present only in kamailio. Does it really
work
in opensips as expected?


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Re: [OpenSIPS-Users] check_source_address()

[Mark Farmer]

Thanks Ben, that seems to be the case. I will update the list with the
results.

Mark.


On Fri, 26 Apr 2019 at 14:09, Ben Newlin  wrote:

> It would cause the issue if they are sending all requests to that domain,
> including sequential requests like re-invite, and ignoring the Contact
> provided in the 200 OK. That is not correct according to RFC 3261, but I
> have seen many carriers do this.
>
>
>
> Ben Newlin
>
>
>
> *From: *Users  on behalf of Mark Farmer
> 
> *Reply-To: *OpenSIPS users mailling list 
> *Date: *Friday, April 26, 2019 at 8:59 AM
> *To: *OpenSIPS users mailling list 
> *Subject: *Re: [OpenSIPS-Users] check_source_address()
>
>
>
> Thank you, that makes sense now. I will keep that in mind for the future.
>
> In the meantime I have raised a query with our provider.
>
>
>
> Additionally, I realised this morning that at our request, our provider is
> sending calls to us via a domain name instead of an IP. Would that likely
> cause the issue even if they are using RFC 3261? I have asked for it to be
> removed.
>
>
>
> Best regards
>
> Mark.
>
>
>
>
>
>
>
> On Thu, 25 Apr 2019 at 16:50, Liviu Chircu  wrote:
>
> On 25.04.2019 17:11, Mark Farmer wrote:
>
> Thanks so much for helping with this.
>
>
>
> I have applied the suggested config but the result is the same. OpenSIPS
> routes the RE-INVITE to itself and it never gets routed back to the
> Asterisk box.
>
> If the 2nd Route header in the RE-INVITE is the IP of the other interface
> - will that not always be the case? It's as though the 2nd Route header
> needs to be changed to have the IP of the Asterisk server.
>
>
>
> Sanitized RE-INVITE from provider:
>
>
>
> INVITE sip:aster...@my.host.name:5060 SIP/2.0
>
>
>
> If OpenSIPS identifies "my.host.name:5060" as a local domain, this will
> screw up the routing,
> as it will go from loose (RFC 3261) to strict (old, deprecated RFC 2543
> mechanism).  Notice how
> its not preserving the R-URI when it routes to itself as should happen
> with RFC 3261 routing,
> because it has fallen back to RFC 2543 routing.
>
> Your provider needs to follow RFC 3261 and use as Re-INVITE Request-URI
> the exact Contact
> advertised by the caller: 
> , and not confuse your routing engine
> with a random target such as: INVITE sip:aster...@my.host.name:5060.
>
> --
>
> Liviu Chircu
>
> OpenSIPS Developer
>
> http://www.opensips-solutions.com
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>
> --
>
> Mark Farmer
> farm...@gmail.com
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>


-- 
Mark Farmer
farm...@gmail.com
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Re: [OpenSIPS-Users] opensips and multiple rtpengine instances

[Johan De Clercq]

I remember that somebody from smartvox  opened an issue on this (it was
either Pete Kelly or John Quick).

Pete,John
do you recall ?

Op vr 26 apr. 2019 om 15:15 schreef Vitalii Aleksandrov <
vitalik.v...@gmail.com>:

> Hello opensips users,
>
> has anybody tried to configure opensips with multiple rtpengine sockets?
>
> Logically thinking offer/answer from initial INVITE/OK and from all
> in-dialog messages must go to the same instance and opensips can load
> balance only initial INVITEs.
> I briefly compared opensips/rtpengine and kamaio/rtpengine modules and
> found that this logic is present only in kamailio. Does it really work
> in opensips as expected?
>
>
> ___
> 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

[OpenSIPS-Users] opensips and multiple rtpengine instances

[Vitalii Aleksandrov]

Hello opensips users,

has anybody tried to configure opensips with multiple rtpengine sockets?

Logically thinking offer/answer from initial INVITE/OK and from all 
in-dialog messages must go to the same instance and opensips can load 
balance only initial INVITEs.
I briefly compared opensips/rtpengine and kamaio/rtpengine modules and 
found that this logic is present only in kamailio. Does it really work 
in opensips as expected?



___
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Re: [OpenSIPS-Users] check_source_address()

[Ben Newlin]

It would cause the issue if they are sending all requests to that domain, 
including sequential requests like re-invite, and ignoring the Contact provided 
in the 200 OK. That is not correct according to RFC 3261, but I have seen many 
carriers do this.

Ben Newlin

From: Users  on behalf of Mark Farmer 

Reply-To: OpenSIPS users mailling list 
Date: Friday, April 26, 2019 at 8:59 AM
To: OpenSIPS users mailling list 
Subject: Re: [OpenSIPS-Users] check_source_address()

Thank you, that makes sense now. I will keep that in mind for the future.
In the meantime I have raised a query with our provider.

Additionally, I realised this morning that at our request, our provider is 
sending calls to us via a domain name instead of an IP. Would that likely cause 
the issue even if they are using RFC 3261? I have asked for it to be removed.

Best regards
Mark.



On Thu, 25 Apr 2019 at 16:50, Liviu Chircu 
mailto:li...@opensips.org>> wrote:
On 25.04.2019 17:11, Mark Farmer wrote:
Thanks so much for helping with this.

I have applied the suggested config but the result is the same. OpenSIPS routes 
the RE-INVITE to itself and it never gets routed back to the Asterisk box.
If the 2nd Route header in the RE-INVITE is the IP of the other interface - 
will that not always be the case? It's as though the 2nd Route header needs to 
be changed to have the IP of the Asterisk server.

Sanitized RE-INVITE from provider:

INVITE sip:aster...@my.host.name:5060 
SIP/2.0


If OpenSIPS identifies "my.host.name:5060" as a local 
domain, this will screw up the routing,
as it will go from loose (RFC 3261) to strict (old, deprecated RFC 2543 
mechanism).  Notice how
its not preserving the R-URI when it routes to itself as should happen with RFC 
3261 routing,
because it has fallen back to RFC 2543 routing.

Your provider needs to follow RFC 3261 and use as Re-INVITE Request-URI the 
exact Contact
advertised by the caller: 
, and not 
confuse your routing engine
with a random target such as: INVITE 
sip:aster...@my.host.name:5060.

--

Liviu Chircu

OpenSIPS Developer

http://www.opensips-solutions.com
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--
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Re: [OpenSIPS-Users] check_source_address()

[Mark Farmer]

Thank you, that makes sense now. I will keep that in mind for the future.
In the meantime I have raised a query with our provider.

Additionally, I realised this morning that at our request, our provider is
sending calls to us via a domain name instead of an IP. Would that likely
cause the issue even if they are using RFC 3261? I have asked for it to be
removed.

Best regards
Mark.



On Thu, 25 Apr 2019 at 16:50, Liviu Chircu  wrote:

> On 25.04.2019 17:11, Mark Farmer wrote:
>
> Thanks so much for helping with this.
>
> I have applied the suggested config but the result is the same. OpenSIPS
> routes the RE-INVITE to itself and it never gets routed back to the
> Asterisk box.
> If the 2nd Route header in the RE-INVITE is the IP of the other interface
> - will that not always be the case? It's as though the 2nd Route header
> needs to be changed to have the IP of the Asterisk server.
>
> Sanitized RE-INVITE from provider:
>
> INVITE sip:aster...@my.host.name:5060 SIP/2.0
>
> If OpenSIPS identifies "my.host.name:5060" as a local domain, this will
> screw up the routing,
> as it will go from loose (RFC 3261) to strict (old, deprecated RFC 2543
> mechanism).  Notice how
> its not preserving the R-URI when it routes to itself as should happen
> with RFC 3261 routing,
> because it has fallen back to RFC 2543 routing.
>
> Your provider needs to follow RFC 3261 and use as Re-INVITE Request-URI
> the exact Contact
> advertised by the caller: 
> , and not confuse your routing engine
> with a random target such as: INVITE sip:aster...@my.host.name:5060.
>
> --
> Liviu Chircu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>


-- 
Mark Farmer
farm...@gmail.com
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