[SR-Users] SIP Recorder

[marius zbihlei]

On 01/26/2011 04:07 PM, Jeremya wrote:
> Someone correct me if I'm wrong, but I've seen enough examples of
> out-of-dialog requests (e.g. BYE) not using the record route to wonder
> if this is in fact required for a new dialog.
>
>    

Hello

You seem to misunderstand some notions. First of all, RR will affect 
future in-dialog requests (The Record-Route header field is used by 
proxies to indicate that they wish to remain in the message path for 
requests send within a dialog.) Also, an out -of -dialog BYE makes 
little sense (as opposed to CANCEL) because the BYE is a specific method 
to close a dialog started with the INVITE request. The proxy should 
forward the request, but a UAS/SBC will return a 481.

Again, a new dialog will never use previous RR headers from other 
dialogs.(if I understand what you are saying).

Marius
> I've managed this by setting outbound proxy, but a general rule would help.
>
> marius zbihlei wrote:
>    
>> On 01/26/2011 03:51 PM, Danny Dias wrote:
>>      
>>> Media NEVER goes through a Proxy core...the question is, how should i
>>> record conversations when the calls are all passing through a sip
>>> proxy? some lights will be enough for me :)
>>>
>>>
>>>        
>> Hello,
>>
>> Use Record-Route headers to force in-dialog requests to have the same
>> path as the original (also you might want to the a look to Path header
>> for REGISTER requests). This will solve the signaling part For Media,
>> I think rtpproxy module will achieve what you want (ignore NAT -
>> basically all you need is to re-write some media attributes in the
>> sdp). The rtpproxy daemon will also be needed.
>>
>> Cheers,
>>
>> Marius
>>      
>>> 2011/1/26 Jeremya<jeremy at electrosilk.net>:
>>>
>>>        
>>>> Whoops! some SIP traffic IS peer-to-peer.
>>>>
>>>> Jeremya wrote:
>>>>
>>>> Danny Dias wrote:
>>>>
>>>>
>>>> Hello my friends,
>>>>
>>>> I have a requeriment, which indicates that i have to record every SIP
>>>> conversation between peers (also for callings to the PSTN); the
>>>> recording server will be built for our company following this
>>>> requeriments (also requested for the client):
>>>>
>>>> My doubt is: How can i handle sip conversations recording when all the
>>>> calls are passing through a Proxy Server? I do understand that the
>>>> media is always peer to peer and the signaling goes through the Proxy,
>>>> but in this case the media not only has to pass between the peers
>>>> because it must be recorded.
>>>>
>>>> How should i handle this?
>>>>
>>>> _______________________________________________
>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>> sr-users at lists.sip-router.org
>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>>
>>>>
>>>> some media is not peer-to-peer. Especially stuff like BYE and NOTIFY.
>>>> Then it is direct to the originator contact address.
>>>>
>>>> Unless you have both ends set up correctly, or you have 'adjusted' the
>>>> SIP traffic, then some stuff may be lost.
>>>>
>>>> _______________________________________________
>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>> sr-users at lists.sip-router.org
>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>>
>>>> _______________________________________________
>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>> sr-users at lists.sip-router.org
>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>>
>>>>
>>>>          
>>>
>>>
>>>        
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>      
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>    

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