Thanks Jeremya, but it's a requeriment from the client to record the calls through an external server and not with rtpproxys, my question is how the media should be handled in order to record the conversations if the server is external?<div>
<br></div><div>Signaling: Phone_A <---> Proxy <---> Phone_B<br><br><div>Media: Phone_A <---> SIP RECORDER <---> Phone_B (Changing the SDP headers to send RTP to the IP of the SIP RECORDER). The main problem is that the recording must be made in ACTIVE way, it means, we should record (IN+OUT) in Phone A, and the same in B, 2 recording for each call...the customer says that it's working now in his arquitecture (its analog), and we made the same with the IP technology...resuming: with a sip recorder in the middle of the media should work right?</div>
<div><br></div><br><div class="gmail_quote">2011/1/26 Jeremya <span dir="ltr"><<a href="mailto:jeremy@electrosilk.net">jeremy@electrosilk.net</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
Someone correct me if I'm wrong, but I've seen enough examples of<br>
out-of-dialog requests (e.g. BYE) not using the record route to wonder<br>
if this is in fact required for a new dialog.<br>
<br>
I've managed this by setting outbound proxy, but a general rule would help.<br>
<div><div></div><div class="h5"><br>
marius zbihlei wrote:<br>
> On 01/26/2011 03:51 PM, Danny Dias wrote:<br>
>> Media NEVER goes through a Proxy core...the question is, how should i<br>
>> record conversations when the calls are all passing through a sip<br>
>> proxy? some lights will be enough for me :)<br>
>><br>
>><br>
><br>
> Hello,<br>
><br>
> Use Record-Route headers to force in-dialog requests to have the same<br>
> path as the original (also you might want to the a look to Path header<br>
> for REGISTER requests). This will solve the signaling part For Media,<br>
> I think rtpproxy module will achieve what you want (ignore NAT -<br>
> basically all you need is to re-write some media attributes in the<br>
> sdp). The rtpproxy daemon will also be needed.<br>
><br>
> Cheers,<br>
><br>
> Marius<br>
>> 2011/1/26 Jeremya<<a href="mailto:jeremy@electrosilk.net">jeremy@electrosilk.net</a>>:<br>
>><br>
>>> Whoops! some SIP traffic IS peer-to-peer.<br>
>>><br>
>>> Jeremya wrote:<br>
>>><br>
>>> Danny Dias wrote:<br>
>>><br>
>>><br>
>>> Hello my friends,<br>
>>><br>
>>> I have a requeriment, which indicates that i have to record every SIP<br>
>>> conversation between peers (also for callings to the PSTN); the<br>
>>> recording server will be built for our company following this<br>
>>> requeriments (also requested for the client):<br>
>>><br>
>>> My doubt is: How can i handle sip conversations recording when all the<br>
>>> calls are passing through a Proxy Server? I do understand that the<br>
>>> media is always peer to peer and the signaling goes through the Proxy,<br>
>>> but in this case the media not only has to pass between the peers<br>
>>> because it must be recorded.<br>
>>><br>
>>> How should i handle this?<br>
>>><br>
>>> _______________________________________________<br>
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
>>> <a href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br>
>>> <a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
>>><br>
>>><br>
>>><br>
>>> some media is not peer-to-peer. Especially stuff like BYE and NOTIFY.<br>
>>> Then it is direct to the originator contact address.<br>
>>><br>
>>> Unless you have both ends set up correctly, or you have 'adjusted' the<br>
>>> SIP traffic, then some stuff may be lost.<br>
>>><br>
>>> _______________________________________________<br>
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
>>> <a href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br>
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>>><br>
>>><br>
>>> _______________________________________________<br>
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
>>> <a href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br>
>>> <a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
>>><br>
>>><br>
>>><br>
>><br>
>><br>
>><br>
><br>
><br>
> _______________________________________________<br>
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
> <a href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br>
> <a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
<br>
<br>
_______________________________________________<br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br>Ing. Danny Dias<br><a href="http://www.DannTEL.net" target="_blank">www.DannTEL.net</a><br>
</div>