[OpenSIPS-Users] OpenSIPS/RTpProxy BridgeMode after failure route

Thu Oct 9 17:42:58 CEST 2014

Hello Răzvan & Salman,

Thank you for your responses.

I was able to fix it by moving rtpproxy_offer to branch route instead of
having it in the main route. In failure route, I only needed to do unforce.

Regards,
Ali Pey

On Thu, Oct 9, 2014 at 3:31 AM, Răzvan Crainea <razvan at opensips.org> wrote:

>  Hi, Ali!
>
> For the initial branch (in request route) are you using engage_rtpproxy()?
> If so, try to use rtpproxy_offer().
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutions <https://contactmonkey.com/api/v1/tracker?cm_session=f72d879b-e536-4227-a59d-8a6c4b8ba6e9&cm_type=link&cm_link=1443c182-b295-4010-9767-59bc588bef8a&cm_destination=http://www.opensips-solutions.com>www.opensips-solutions.com <https://contactmonkey.com/api/v1/tracker?cm_session=f72d879b-e536-4227-a59d-8a6c4b8ba6e9&cm_type=link&cm_link=4fe71057-5c3d-4ff5-89ac-a63e63631bb1&cm_destination=http://www.opensips-solutions.com>
>
> On 10/09/2014 12:06 AM, Ali Pey wrote:
>
>  Hello Salman,
>
>  Can you please elaborate on how you got this working? I have the same
> problem and can't get it to work.
>
>  In failure route, I do:
> unforce_rtp_proxy()
>  Then when I have a new destination, I do:
> rtpproxy_offer("rocie");
>
>  However, I end up with messed up SDP, in my second invite. It doesn't
> remove the old IP addresses and only adds the IP addresses again:
>  o=Sonus_UAC 9216 20203 IN IP4 10.160.11.16210.160.11.162a Capabilities
> c=IN IP4 10.160.11.16210.160.11.162udio 2311822970AVP 0 8 100
>
>
>  Please let me know how I can fix this.
>
>  Thanks.
>
>
> On Mon, Jan 6, 2014 at 10:26 AM, Salman Zafar <msalman212 at gmail.com>
> wrote:
>
>>  Hi Razvan,
>>          I got it working without branching, after banging head a lot I
>> got to learn unforcing drops the media ports for previous rtpproxy
>> offer/answer and after that directing the new flow though rtpproxy flags,IP
>> media works. I am able to traverse from eternal to internal play media and
>> then on failure do external to external with media flowing between public
>> interfaces. Just wondering if you know this method or certify.
>>
>>
>>
>> On Mon, Jan 6, 2014 at 4:35 PM, Răzvan Crainea <razvan at opensips.org>
>> wrote:
>>
>>> Hi, Salman!
>>>
>>> The sockets used by RTPProxy are created when the session is started
>>> (the first offer) and cannot be updated afterwards. Therefore the only
>>> solution I can see is to configure a per branch scenario, as you mentioned.
>>>
>>> Best regards,
>>>
>>> Razvan Crainea
>>> OpenSIPS Core Developer
>>> http://www.opensips-solutions.com
>>> <https://contactmonkey.com/api/v1/tracker?cm_session=66f49ebf-f052-47c3-adee-bf8dd17afa5d&cm_type=link&cm_link=c1574c01-908b-4910-aaff-83f9f8f63efd&cm_destination=http://www.opensips-solutions.com>
>>>
>>>
>>> On 12/30/2013 01:11 PM, Salman Zafar wrote:
>>>
>>>>  Hi,
>>>>     I have a scenario of playing media at a private-ip media server and
>>>> send BUSY, next in failure route bridge call to a public IP. (SIP to
>>>> SIP).
>>>>
>>>> So the scenario is as follows:
>>>>
>>>> UA(Phone1) -> OpenSIPS/RTpProxy(ei) -> Media-Server (Private IP) -> BUSY
>>>> -> OpenSIPS(failure route) -> RTpProxy(ee) -> lookup -> (UA Phone2)
>>>>
>>>> Now the problem is RtpProxy is being offered (EI flags) in first case
>>>> where routing to Media sever at private IP, after failure it is again
>>>> used with (EE flags), also in corresponding replies.
>>>>
>>>> The second time RTpProxy does not effect SDP c= and ports in a way to
>>>> build media communication. SDP fix directly does not effect rtp ports.
>>>>
>>>> Is there any way of using RtpProxy differently in fail-over, or I have
>>>> to go for rtpproxy per branch?.
>>>>
>>>>
>>>> Thanks in advance.
>>>>
>>>> --
>>>> Regards
>>>>
>>>> Salman
>>>>
>>>>
>>>>
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>>>>
>>>>
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>>
>>
>>
>>  --
>> Regards
>>
>>  M. Salman Zafar
>>
>> VoIP Professional
>>
>>
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