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?
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@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.
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@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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 :)
2011/1/26 Jeremya jeremy@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@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@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Actually - being pedantic - some proxy cores have a co-located media server. e.g. rtpproxy.
The problem is getting ALL SIP traffic to run through the same SIP proxy and/or proxies. It usually happens but needs careful attention.
As regards simple accounting, the Kamailio/SER system has full call accounting for INVITE and BYE and a few others.
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 :)
2011/1/26 Jeremya jeremy@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@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@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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 Jeremyajeremy@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@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@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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.
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 Jeremyajeremy@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@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@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@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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?
Signaling: Phone_A <---> Proxy <---> Phone_B
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?
2011/1/26 Jeremya jeremy@electrosilk.net
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.
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 Jeremyajeremy@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@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@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@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@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Danny Dias ing.diasdanny@gmail.com escribió:
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?
Signaling: Phone_A <---> Proxy <---> Phone_B
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?
2 ways of doing that:
a) Signaling: A <-> Proxy <-> B2BUA (recorder) <-> B Media: A <-> B2BUA <-> B
b) Prefered way Signaling: A <-> Proxy <-> B Media: A<-> RTPPROXY <-> B
At the end, both solutions are THE SAME, what you do is to tell A to send media to the B2BUA or the RTPPRoxy.
As a matters of scale ... b) solution is the best one.
Also, another things to take into account are:
1- Transcoding issues (RTPPRoxy does not do transconding, not easly) 2- Secured RTP (ZRTP, SRTP, etc.) 3- LAG in audio.
---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program.
Many thanks Jaremya,
The main problem is that both terminals, SHALL (required and must not be changed, because of standards of EUROCAE ED-137 Part3) initiate a session with the recorder server (a commercial one, can't use Asterisk for my disgrace) sending INVITE and receiving the subsequent responses from sip recording server to stablish the session with it...after this, when the media starts to go directly peer to peer (the normal call), the terminals (specials ones) must summarize the IN+OUT audio to the recording server and through rtsp the media should be recorded...it's weird, but thats the requirement :S
i mean....
signaling: A---->PROXY---->B (the normal procedure)
At the same time, this must be done: (I'm not sure how to do this...the proxy could be out of this or not, not sure :()
A ---INVITE---> SIP_PROXY ---INVITE---> SIP_RECORDER B ---INVITE---> SIP_RECORDER --INVITE--> SIP_RECORDER
Then, The audio will go directly from A to B (because of the normal procedures), and also, A and B, will summarize IN+OUT on each site and send this result through RTSP to the recording server (this is not important to the proxy righ not)...My real doubt is how to stablish the session between the peers A and B to the recording server through the Proxy and also (at the same time) continue with the normal flow of the call (invite from a to b, 200 ok viceversa etc etc...)
Should i use some function like t_replicate to send 2 invites like this:
A --INVITE--> PROXY --INVITE--> B . . INVITE . RECORDER SERVER
But the problem here is that the session between A and PROXY would be OK, but i can't see the way how B should send INVITE to the recorder server..
I hope to be clear on my problem :( and i know it looks very weird, but it's the requirement of the document mentioned above
Thanks in advance!!!
2011/1/26 rabs@dimension-virtual.com
Danny Dias ing.diasdanny@gmail.com escribió:
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?
Signaling: Phone_A <---> Proxy <---> Phone_B
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?
2 ways of doing that:
a) Signaling: A <-> Proxy <-> B2BUA (recorder) <-> B Media: A <-> B2BUA <-> B
b) Prefered way Signaling: A <-> Proxy <-> B Media: A<-> RTPPROXY <-> B
At the end, both solutions are THE SAME, what you do is to tell A to send media to the B2BUA or the RTPPRoxy.
As a matters of scale ... b) solution is the best one.
Also, another things to take into account are:
1- Transcoding issues (RTPPRoxy does not do transconding, not easly) 2- Secured RTP (ZRTP, SRTP, etc.) 3- LAG in audio.
This message was sent using IMP, the Internet Messaging Program.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
OOps, made a mistake on tipping.....take a look down please...
2011/1/26 Danny Dias ing.diasdanny@gmail.com
Many thanks Jaremya,
The main problem is that both terminals, SHALL (required and must not be changed, because of standards of EUROCAE ED-137 Part3) initiate a session with the recorder server (a commercial one, can't use Asterisk for my disgrace) sending INVITE and receiving the subsequent responses from sip recording server to stablish the session with it...after this, when the media starts to go directly peer to peer (the normal call), the terminals (specials ones) must summarize the IN+OUT audio to the recording server and through rtsp the media should be recorded...it's weird, but thats the requirement :S
i mean....
signaling: A---->PROXY---->B (the normal procedure)
At the same time, this must be done: (I'm not sure how to do this...the proxy could be out of this or not, not sure :()
A ---INVITE---> SIP_PROXY ---INVITE---> SIP_RECORDER B ---INVITE---> SIP_RECORDER --INVITE--> SIP_RECORDER
B ---INVITE---> SIP_PROXY ---INVITE---> SIP_RECORDER
Then, The audio will go directly from A to B (because of the normal procedures), and also, A and B, will summarize IN+OUT on each site and send this result through RTSP to the recording server (this is not important to the proxy righ not)...My real doubt is how to stablish the session between the peers A and B to the recording server through the Proxy and also (at the same time) continue with the normal flow of the call (invite from a to b, 200 ok viceversa etc etc...)
Should i use some function like t_replicate to send 2 invites like this:
A --INVITE--> PROXY --INVITE--> B . . INVITE . RECORDER SERVER
But the problem here is that the session between A and PROXY would be OK, but i can't see the way how B should send INVITE to the recorder server..
I hope to be clear on my problem :( and i know it looks very weird, but it's the requirement of the document mentioned above
Thanks in advance!!!
2011/1/26 rabs@dimension-virtual.com
Danny Dias ing.diasdanny@gmail.com escribió:
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?
Signaling: Phone_A <---> Proxy <---> Phone_B
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?
2 ways of doing that:
a) Signaling: A <-> Proxy <-> B2BUA (recorder) <-> B Media: A <-> B2BUA <-> B
b) Prefered way Signaling: A <-> Proxy <-> B Media: A<-> RTPPROXY <-> B
At the end, both solutions are THE SAME, what you do is to tell A to send media to the B2BUA or the RTPPRoxy.
As a matters of scale ... b) solution is the best one.
Also, another things to take into account are:
1- Transcoding issues (RTPPRoxy does not do transconding, not easly) 2- Secured RTP (ZRTP, SRTP, etc.) 3- LAG in audio.
This message was sent using IMP, the Internet Messaging Program.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
As said, SER is not the best vehicle for that as it in its SIP proxy definition does not process media. SEMS does this (recording, mixing, storing....) quite decently. I'm just wondering what is the reason that Asterisk can't be used -- perhaps SEMS would fail that criteria as well.
-jiri
On 1/26/11 6:43 PM, Danny Dias wrote:
Many thanks Jaremya,
The main problem is that both terminals, SHALL (required and must not be changed, because of standards of EUROCAE ED-137 Part3) initiate a session with the recorder server (a commercial one, can't use Asterisk for my disgrace) sending INVITE and receiving the subsequent responses from sip recording server to stablish the session with it...after this, when the media starts to go directly peer to peer (the normal call), the terminals (specials ones) must summarize the IN+OUT audio to the recording server and through rtsp the media should be recorded...it's weird, but thats the requirement :S
i mean....
signaling: A---->PROXY---->B (the normal procedure)
At the same time, this must be done: (I'm not sure how to do this...the proxy could be out of this or not, not sure :()
A ---INVITE---> SIP_PROXY ---INVITE---> SIP_RECORDER B ---INVITE---> SIP_RECORDER --INVITE--> SIP_RECORDER
Then, The audio will go directly from A to B (because of the normal procedures), and also, A and B, will summarize IN+OUT on each site and send this result through RTSP to the recording server (this is not important to the proxy righ not)...My real doubt is how to stablish the session between the peers A and B to the recording server through the Proxy and also (at the same time) continue with the normal flow of the call (invite from a to b, 200 ok viceversa etc etc...)
Should i use some function like t_replicate to send 2 invites like this:
A --INVITE--> PROXY --INVITE--> B . . INVITE . RECORDER SERVER
But the problem here is that the session between A and PROXY would be OK, but i can't see the way how B should send INVITE to the recorder server..
I hope to be clear on my problem :( and i know it looks very weird, but it's the requirement of the document mentioned above
Thanks in advance!!!
2011/1/26 <rabs@dimension-virtual.com mailto:rabs@dimension-virtual.com>
Danny Dias <ing.diasdanny@gmail.com <mailto:ing.diasdanny@gmail.com>> escribió: 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? Signaling: Phone_A <---> Proxy <---> Phone_B 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? 2 ways of doing that: a) Signaling: A <-> Proxy <-> B2BUA (recorder) <-> B Media: A <-> B2BUA <-> B b) Prefered way Signaling: A <-> Proxy <-> B Media: A<-> RTPPROXY <-> B At the end, both solutions are THE SAME, what you do is to tell A to send media to the B2BUA or the RTPPRoxy. As a matters of scale ... b) solution is the best one. Also, another things to take into account are: 1- Transcoding issues (RTPPRoxy does not do transconding, not easly) 2- Secured RTP (ZRTP, SRTP, etc.) 3- LAG in audio. ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Danny Dias ing.diasdanny@gmail.com escribió:
Many thanks Jaremya,
The main problem is that both terminals, SHALL (required and must not be changed, because of standards of EUROCAE ED-137 Part3) initiate a session with the recorder server (a commercial one, can't use Asterisk for my disgrace) sending INVITE and receiving the subsequent responses from sip recording server to stablish the session with it...after this, when the media starts to go directly peer to peer (the normal call), the terminals (specials ones) must summarize the IN+OUT audio to the recording server and through rtsp the media should be recorded...it's weird, but thats the requirement :S
i mean....
signaling: A---->PROXY---->B (the normal procedure)
At the same time, this must be done: (I'm not sure how to do this...the proxy could be out of this or not, not sure :()
A ---INVITE---> SIP_PROXY ---INVITE---> SIP_RECORDER B ---INVITE---> SIP_RECORDER --INVITE--> SIP_RECORDER
Then, The audio will go directly from A to B (because of the normal procedures), and also, A and B, will summarize IN+OUT on each site and send this result through RTSP to the recording server (this is not important to the proxy righ not)...My real doubt is how to stablish the session between the peers A and B to the recording server through the Proxy and also (at the same time) continue with the normal flow of the call (invite from a to b, 200 ok viceversa etc etc...)
Should i use some function like t_replicate to send 2 invites like this:
A --INVITE--> PROXY --INVITE--> B . . INVITE . RECORDER SERVER
But the problem here is that the session between A and PROXY would be OK, but i can't see the way how B should send INVITE to the recorder server..
I hope to be clear on my problem :( and i know it looks very weird, but it's the requirement of the document mentioned above
But tha's not a SIP flow for a call stablishment ... it seems more like a conference service, than a call service.
How does B know that A wants to talk with him? ... It doesn't know
Also, no matter if they are "special" SIP terminals, because you say that the will "sumarize IN+OUT and send it to the record server" ... dear sir ... that's not SIP compliant at all!
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2011/1/26 Danny Dias ing.diasdanny@gmail.com:
i mean.... signaling: A---->PROXY---->B (the normal procedure) At the same time, this must be done: (I'm not sure how to do this...the proxy could be out of this or not, not sure :() A ---INVITE---> SIP_PROXY ---INVITE---> SIP_RECORDER B ---INVITE---> SIP_RECORDER --INVITE--> SIP_RECORDER
Hi, is such SIP_PROXY an instance of Kamailio/SER?
Hi Iñaki,
2011/1/27 Iñaki Baz Castillo ibc@aliax.net
2011/1/26 Danny Dias ing.diasdanny@gmail.com:
i mean.... signaling: A---->PROXY---->B (the normal procedure) At the same time, this must be done: (I'm not sure how to do this...the proxy could be out of this or not, not sure :() A ---INVITE---> SIP_PROXY ---INVITE---> SIP_RECORDER B ---INVITE---> SIP_RECORDER --INVITE--> SIP_RECORDER
B ---INVITE---> SIP_PROXY --INVITE--> SIP_RECORDER
Each peer must send an INVITE to the sip_recorder server, to stablish a session with it...
Hi, is such SIP_PROXY an instance of Kamailio/SER?
YEs, the proxys are Kamailio.
-- Iñaki Baz Castillo ibc@aliax.net
Danny Dias ing.diasdanny@gmail.com escribió:
Hi Iñaki,
2011/1/27 Iñaki Baz Castillo ibc@aliax.net
2011/1/26 Danny Dias ing.diasdanny@gmail.com:
i mean.... signaling: A---->PROXY---->B (the normal procedure) At the same time, this must be done: (I'm not sure how to do this...the proxy could be out of this or not, not sure :() A ---INVITE---> SIP_PROXY ---INVITE---> SIP_RECORDER B ---INVITE---> SIP_RECORDER --INVITE--> SIP_RECORDER
B ---INVITE---> SIP_PROXY --INVITE--> SIP_RECORDER
Each peer must send an INVITE to the sip_recorder server, to stablish a session with it...
So ... if each UAC send an INVITE to the SIP-RECODER ... where is the problem? ... the RTP will be from UAC A to SIP-RECORDER .. that's all. Nothing strange should be done
Best regards
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o rabs@dimension-virtual.com on 01/27/2011 01:13 PM:
Danny Dias ing.diasdanny@gmail.com escribió:
Hi Iñaki,
2011/1/27 Iñaki Baz Castillo ibc@aliax.net
2011/1/26 Danny Dias ing.diasdanny@gmail.com:
i mean.... signaling: A---->PROXY---->B (the normal procedure) At the same time, this must be done: (I'm not sure how to do
this...the
proxy could be out of this or not, not sure :() A ---INVITE---> SIP_PROXY ---INVITE---> SIP_RECORDER B ---INVITE---> SIP_RECORDER --INVITE--> SIP_RECORDER
B ---INVITE---> SIP_PROXY --INVITE--> SIP_RECORDER
Each peer must send an INVITE to the sip_recorder server, to stablish a session with it...
So ... if each UAC send an INVITE to the SIP-RECODER ... where is the problem? ... the RTP will be from UAC A to SIP-RECORDER .. that's all.
no, I think what he wants is in the proxy to intercept the call which is from UAC A to UAC B, and place two additional calls to his SIP-RECORDER, which should transmit both signaling and RTP to the SIP-RECORDER.
To the OP: this is IMO not possible to do properly in a SIP proxy, even if its as flexible as s-r; while you can replicate the initial INVITE to the SIP-RECORDER, and even change the RTP media address in the SDP in order to make RTP go through rtpproxy, which might be modified to send a copy of the traffic to your SIP recorder, I really doubt that you can change the 200 to establish another INVITE to the SIP-RECORDER, let alone that you need to process the responses to the INVITEs to SIp-RECORDER, generate ACKs, handle in-dialog requests (e.g. session timer, hold etc etc).
For this stuff you need a SIP app server/B2BUA/media server, which has two dialogs on each side, and another two dialogs to the SIP-RECORDER. You might possibly be interested in SEMS' sbc module (shameless ad ;), latest development features RTP relay, you might be able to change it to create two more dialogs to SIP-RECORDER, using the original signaling, and relay the RTP also there. If that doesn't suit you, possibly you could have a look at implementing this using pjsip, freeswitch, asterisk (maybe in that order, depending on how transparent the thing in the middle should be). An alternative is to use s-r proxy and rtpproxy, and add some custom app server (again built possibly with sems/pjsip/freeswitch etc), which somehow gets signaling and RTP traffic from s-r/rtpproxy and sends them into two separate calls which are established to SIP-RECORDER. This type of stuff is usually implemented for LI (which hopefully we all don't like) and has further requirements regarding being transparent etc.
hth Stefan
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 Jeremyajeremy@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@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@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@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@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Danny Dias ing.diasdanny@gmail.com escribió:
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?
Forcing all the RTP traffic to pass throught your rtpproxies, that is, handling all the call as if the wall allways behind a NAT.
Also, you would need a rtpproxy that could record the calls, as rtpproxy does.
Another aproach, is to use a Full B2BUA, that does all the recording.
Best regards
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the SIP proxy server does not see media indeed, you better look at a media server such as SEMS. (iptel.org/sems)
jiri
On 1/26/11 2:41 PM, 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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-
Danny Dias -
Iñaki Baz Castillo -
Jeremya -
Jiri Kuthan -
marius zbihlei -
rabs＠dimension-virtual.com -
Stefan Sayer