[OpenSIPS-Users] Question regarding Federated User Location Cluster

[Jeffrey Zhao]

Dear Team
After reading through below Tutorial, I have some questions regarding Database 
and NoSQL setup model for Federated User Location Cluster.




https://opensips.org/Documentation/Tutorials-Distributed-User-Location-Federation


For example, for formal production system senario, two sites, with two opensips 
nodes for each site, HA mode for each site.
1. Should I deploy MySQL and Cassandra on each node? 4 MySQL instances and 4 
Cassandra on each node?
2. Should I setup db replication among 4 MySQL instances or just standalone 
separated setup?
3. Should I put all 4 Cassandra instances into one cluster?
4. For each HA virtual IP setup, what's the recommended tool, keepalived?




Thanks in advance.


Best wishes,
Jeffrey___
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[OpenSIPS-Users] Siprec contact header

[jofi Y]

Hello All,

I have a question regarding contact header generated by siprec module.
According to rfc7866 (6.1.1):

  The SRC MUST include the "+sip.src" feature tag in the Contact URI,
   defined in this specification as an extension to [RFC3840
], for all
   RSs.


The contact parameter "+sip.src" is not generated when I started recording.
Currently I use Opensips version 2.4.6 and 3.1.1.

Is there an alternative way to generate "+sip.src" when recording session
started?

Many thanks,
Jofi
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Re: [OpenSIPS-Users] How to see outgoing TLS OPTIONS

[Johan De Clercq]

Install Homer with portmirror on switch

On Thu, Jan 21, 2021, 13:31 Mark Farmer  wrote:

> Hi everyone
>
> In a recent thread I learnt that using trace() in local_route after
> appending a Contact header traces the message before the new header is
> actually added which means I don't get to see it.
>
> Is there a way to see the OPTIONS message after the new header is added?
>
> Many thanks
> Mark.
>
>
> On Thu, 12 Mar 2020 at 13:02, Mark Farmer  wrote:
>
>> I think I answered my own question :)
>>
>> Adding a sip_trace() into local_route seems to do the trick :)
>>
>> Mark.
>>
>>
>> On Thu, 12 Mar 2020 at 12:31, Mark Farmer  wrote:
>>
>>> Hi everyone
>>>
>>> I am using the drouting module to make SIP/TLS connections and I need to
>>> be able to capture the outgoing OPTIONS requests generated by drouting.
>>>
>>> I am thinking sip_trace("hep_dst", "d"); but where would I need to do
>>> that?
>>>
>>> Is there a better way?
>>>
>>> OpenSIPS 2.4
>>>
>>> Many thanks!
>>> Mark.
>>>
>>>
>>
>> --
>> Mark Farmer
>> farm...@gmail.com
>>
>
>
> --
> Mark Farmer
> farm...@gmail.com
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Re: [OpenSIPS-Users] How to see outgoing TLS OPTIONS

[Mark Farmer]

Hi everyone

In a recent thread I learnt that using trace() in local_route after
appending a Contact header traces the message before the new header is
actually added which means I don't get to see it.

Is there a way to see the OPTIONS message after the new header is added?

Many thanks
Mark.


On Thu, 12 Mar 2020 at 13:02, Mark Farmer  wrote:

> I think I answered my own question :)
>
> Adding a sip_trace() into local_route seems to do the trick :)
>
> Mark.
>
>
> On Thu, 12 Mar 2020 at 12:31, Mark Farmer  wrote:
>
>> Hi everyone
>>
>> I am using the drouting module to make SIP/TLS connections and I need to
>> be able to capture the outgoing OPTIONS requests generated by drouting.
>>
>> I am thinking sip_trace("hep_dst", "d"); but where would I need to do
>> that?
>>
>> Is there a better way?
>>
>> OpenSIPS 2.4
>>
>> Many thanks!
>> Mark.
>>
>>
>
> --
> Mark Farmer
> farm...@gmail.com
>


-- 
Mark Farmer
farm...@gmail.com
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Re: [OpenSIPS-Users] Mediaproxy configuration

[Johan De Clercq]

I totally agree with the rtpengine suggestion

Outlook voor iOS downloaden

Van: Users  namens John Quick 

Verzonden: Thursday, January 21, 2021 10:40:18 AM
Aan: users@lists.opensips.org 
Onderwerp: Re: [OpenSIPS-Users] Mediaproxy configuration

Mark,

I recommend using rtpproxy (or possibly rtpengine) rather than mediaproxy
for your situation.
You need the address in the SDP to be the public IP 4x.xxx.xxx.xxx when it
is sending packets to the UAC but you need it to use its LAN address when
sending to the Asterisk server.
This is what bridge mode (or bridging mode) is used for, although the last
time I built a solution like this I didn't use bridge mode and instead
passed the relevant IP address as an argument when calling the rtpproxy
activation functions. Unfortunately, the latter approach means your
opensips.cfg script will need to be much more complicated.

I suspect your problem when using mediaproxy and advertised_ip =
4x.xxx.xxx.xxx is that it will pass that address to Asterisk in the SDP. In
which case, you might be able to get audio if you look at the network route
Asterisk would use to reach/connect to 4x.xxx.xxx.xxx and make sure the
mediaproxy relay is reachable. However, that does not sound like a good
solution to me - much better if Asterisk talks to the relay directly over
the LAN.

John Quick
Smartvox Limited
Web: www.smartvox.co.uk



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Re: [OpenSIPS-Users] Mediaproxy configuration

[John Quick]

Mark,

I recommend using rtpproxy (or possibly rtpengine) rather than mediaproxy
for your situation.
You need the address in the SDP to be the public IP 4x.xxx.xxx.xxx when it
is sending packets to the UAC but you need it to use its LAN address when
sending to the Asterisk server.
This is what bridge mode (or bridging mode) is used for, although the last
time I built a solution like this I didn't use bridge mode and instead
passed the relevant IP address as an argument when calling the rtpproxy
activation functions. Unfortunately, the latter approach means your
opensips.cfg script will need to be much more complicated.

I suspect your problem when using mediaproxy and advertised_ip =
4x.xxx.xxx.xxx is that it will pass that address to Asterisk in the SDP. In
which case, you might be able to get audio if you look at the network route
Asterisk would use to reach/connect to 4x.xxx.xxx.xxx and make sure the
mediaproxy relay is reachable. However, that does not sound like a good
solution to me - much better if Asterisk talks to the relay directly over
the LAN.

John Quick
Smartvox Limited
Web: www.smartvox.co.uk



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