All-
Can we use Asterisk combined with Kamailio as follows:
__________ ___________ | | | | SIP ___| |___ SIP ___| Kamailio |___ SIP | | | rtpproxy | | Asterisk | | | | | | | | | RTP ___| |___ RTP ___| DSP card |___ RTP (G711) |__________| (G711) |___________| (G729, G723, GSM-AMR, EVRC)
We've already implemented an rtpproxy interface to the DSP card, which has its own GbE port. Our question is whether we can we perform some type of basic SIP forwarding or "SIP pass-thru", but still invoke rtpproxy for call setup and tear-down and/or when media attributes for the call change?
We're getting a lot of requests from customers who -- for whatever reasons -- need to use or continue to use Asterisk, but need to also add transcoding, ec, encryption, and other compute-intensive requirements that Asterisk doesn't support (or at least doesn't support at higher capacity or without going unstable).
Thanks.
-Jeff
Hi Jeff!
So you want to do transcoding in rtpproxy using a DSP card? I do not know - better ask on the rtpproxy mailing list (or Maxim directly - I think he has a non-open source solution).
Anyway - why not do the transcoding in Asterisk?
regards klaus
Jeff Brower schrieb:
All-
Can we use Asterisk combined with Kamailio as follows:
__________ ___________ | | | |SIP ___| |___ SIP ___| Kamailio |___ SIP | | | rtpproxy | | Asterisk | | | | | | | | | RTP ___| |___ RTP ___| DSP card |___ RTP (G711) |__________| (G711) |___________| (G729, G723, GSM-AMR, EVRC)
We've already implemented an rtpproxy interface to the DSP card, which has its own GbE port. Our question is whether we can we perform some type of basic SIP forwarding or "SIP pass-thru", but still invoke rtpproxy for call setup and tear-down and/or when media attributes for the call change?
We're getting a lot of requests from customers who -- for whatever reasons -- need to use or continue to use Asterisk, but need to also add transcoding, ec, encryption, and other compute-intensive requirements that Asterisk doesn't support (or at least doesn't support at higher capacity or without going unstable).
Thanks.
-Jeff
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Klaus-
So you want to do transcoding in rtpproxy using a DSP card? I do not know - better ask on the rtpproxy mailing list (or Maxim directly - I think he has a non-open source solution).
Ya we have -- and it works, no problem. We've tested already with Kamailio + rtpproxy.
Anyway - why not do the transcoding in Asterisk?
Because Asterisk is too limited. It can't do enough channels for G729, and doesn't have good options for codecs like EVRC and GSM-AMR.
But anyway my question is about SIP with Kamailio + Asterisk, not RTP. Is there a way that Kamailio can "pass thru" SIP messages from Asterisk? Or does each call have to be relayed; i.e Asterisk sets up a call to Kamailio, then Kamalio sets up a call to the endpoint?
I know that we can get it to work one way or another, but I'm worried about channel capacity if both Asterisk and Kamailio run on the same server. "Duplicating" calls does not seem efficient.
-Jeff
Jeff Brower schrieb:
All-
Can we use Asterisk combined with Kamailio as follows:
__________ ___________ | | | |SIP ___| |___ SIP ___| Kamailio |___ SIP | | | rtpproxy | | Asterisk | | | | | | | | | RTP ___| |___ RTP ___| DSP card |___ RTP (G711) |__________| (G711) |___________| (G729, G723, GSM-AMR, EVRC)
We've already implemented an rtpproxy interface to the DSP card, which has its own GbE port. Our question is whether we can we perform some type of basic SIP forwarding or "SIP pass-thru", but still invoke rtpproxy for call setup and tear-down and/or when media attributes for the call change?
We're getting a lot of requests from customers who -- for whatever reasons -- need to use or continue to use Asterisk, but need to also add transcoding, ec, encryption, and other compute-intensive requirements that Asterisk doesn't support (or at least doesn't support at higher capacity or without going unstable).
Thanks.
-Jeff
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Jeff Brower schrieb:
Klaus-
So you want to do transcoding in rtpproxy using a DSP card? I do not know - better ask on the rtpproxy mailing list (or Maxim directly - I think he has a non-open source solution).
Ya we have -- and it works, no problem. We've tested already with Kamailio + rtpproxy.
Anyway - why not do the transcoding in Asterisk?
Because Asterisk is too limited. It can't do enough channels for G729, and doesn't have good options for codecs like EVRC and GSM-AMR.
But anyway my question is about SIP with Kamailio + Asterisk, not RTP. Is there a way that Kamailio can "pass thru" SIP messages from Asterisk? Or does each call have to be relayed; i.e Asterisk sets up a call to Kamailio, then Kamalio sets up a call to the endpoint?
What is the difference between "pass thru" and relaying? Kamailio is a proxy, that means it receives a SIP message from somebody and sends it (slighty modified) to somebody. This forwarding can be done stateless or transaction statefull.
Unless you use the dialog module, Kamailio does not care about "set-up" calls, so if you have 100000000 millions if calls set up, Kamailio is idle as it only cares about transactions (INVITE, BYE ...), not about ongoing calls.
regards klaus
Klaus-
Jeff Brower schrieb:
Klaus-
So you want to do transcoding in rtpproxy using a DSP card? I do not know - better ask on the rtpproxy mailing list (or Maxim directly - I think he has a non-open source solution).
Ya we have -- and it works, no problem. We've tested already with Kamailio + rtpproxy.
Anyway - why not do the transcoding in Asterisk?
Because Asterisk is too limited. It can't do enough channels for G729, and doesn't have good options for codecs like EVRC and GSM-AMR.
But anyway my question is about SIP with Kamailio + Asterisk, not RTP. Is there a way that Kamailio can "pass thru" SIP messages from Asterisk? Or does each call have to be relayed; i.e Asterisk sets up a call to Kamailio, then Kamalio sets up a call to the endpoint?
What is the difference between "pass thru" and relaying? Kamailio is a proxy, that means it receives a SIP message from somebody and sends it (slighty modified) to somebody. This forwarding can be done stateless or transaction statefull.
Unless you use the dialog module, Kamailio does not care about "set-up" calls, so if you have 100000000 millions if calls set up, Kamailio is idle as it only cares about transactions (INVITE, BYE ...), not about ongoing calls.
Thanks for your reply. Yes you're right... I think we just need to try it stateless and measure the performance.
-Jeff
On Friday 23 October 2009 16:01:41 Jeff Brower wrote:
Klaus-
So you want to do transcoding in rtpproxy using a DSP card? I do not know - better ask on the rtpproxy mailing list (or Maxim directly - I think he has a non-open source solution).
Ya we have -- and it works, no problem. We've tested already with Kamailio
- rtpproxy.
It's possible to know witch DSP card are you using ?
But anyway my question is about SIP with Kamailio + Asterisk, not RTP. Is there a way that Kamailio can "pass thru" SIP messages from Asterisk? Or does each call have to be relayed; i.e Asterisk sets up a call to Kamailio, then Kamalio sets up a call to the endpoint?
Kamailio as SIP proxy, doesn't "setup a call", it just route SIP messages, so by definition it works as "SIP-Passthrought", so what is the problem you have ?
Raúl Alexis-
On Friday 23 October 2009 16:01:41 Jeff Brower wrote:
Klaus-
So you want to do transcoding in rtpproxy using a DSP card? I do not know - better ask on the rtpproxy mailing list (or Maxim directly - I think he has a non-open source solution).
Ya we have -- and it works, no problem. We've tested already with Kamailio
- rtpproxy.
It's possible to know witch DSP card are you using ?
http://www.signalogic.com/sigc5561_ptmc.htm
in PCI and PCIe formats.
But anyway my question is about SIP with Kamailio + Asterisk, not RTP. Is there a way that Kamailio can "pass thru" SIP messages from Asterisk? Or does each call have to be relayed; i.e Asterisk sets up a call to Kamailio, then Kamalio sets up a call to the endpoint?
Kamailio as SIP proxy, doesn't "setup a call", it just route SIP messages, so by definition it works as "SIP-Passthrought", so what is the problem you have ?
Well hopefully no problem and I'm a worry-wart :-) We're very sensitive to call setup and tear-down times... especially we're concerned about running Asterisk and Kamailio on the same server. Doing RTP stuff with a DSP card gives a huge increase in call capacity, but if we can't maintain fast setup and tear-down rates, then we defeat the purpose.
-Jeff
Raúl Alexis Betancor Santana Dimensión Virtual
Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Jeff Brower schrieb:
Raúl Alexis-
On Friday 23 October 2009 16:01:41 Jeff Brower wrote:
Klaus-
So you want to do transcoding in rtpproxy using a DSP card? I do not know - better ask on the rtpproxy mailing list (or Maxim directly - I think he has a non-open source solution).
Ya we have -- and it works, no problem. We've tested already with Kamailio
- rtpproxy.
It's possible to know witch DSP card are you using ?
http://www.signalogic.com/sigc5561_ptmc.htm
in PCI and PCIe formats.
Interesting. How does it work in detail. Does rtpproxy use the DSP card for transcoding, or is rtpproxy bypassed completely and Kamailio talkes with the DSP card directly which works like rtpproxy itself (as I saw the DPS card has a dedicated network inteface)?
regards klaus
Klaus-
Jeff Brower schrieb:
Raúl Alexis-
On Friday 23 October 2009 16:01:41 Jeff Brower wrote:
Klaus-
So you want to do transcoding in rtpproxy using a DSP card? I do not know - better ask on the rtpproxy mailing list (or Maxim directly - I think he has a non-open source solution).
Ya we have -- and it works, no problem. We've tested already with Kamailio
- rtpproxy.
It's possible to know witch DSP card are you using ?
http://www.signalogic.com/sigc5561_ptmc.htm
in PCI and PCIe formats.
Interesting. How does it work in detail. Does rtpproxy use the DSP card for transcoding, or is rtpproxy bypassed completely and Kamailio talkes with the DSP card directly which works like rtpproxy itself (as I saw the DPS card has a dedicated network inteface)?
rtpproxy interfaces with the card. rtpproxy uses the card's GbE for IP addr's instead of the server. So the idea (more or less) is the server handles signaling / SIP and card handles data / RTP. We keep Kamailio away from the card, so it doesn't have to be in the same server.
-Jeff
Jeff Brower schrieb:
Klaus-
Jeff Brower schrieb:
Raúl Alexis-
On Friday 23 October 2009 16:01:41 Jeff Brower wrote:
Klaus-
So you want to do transcoding in rtpproxy using a DSP card? I do not know - better ask on the rtpproxy mailing list (or Maxim directly - I think he has a non-open source solution).
Ya we have -- and it works, no problem. We've tested already with Kamailio
- rtpproxy.
It's possible to know witch DSP card are you using ?
http://www.signalogic.com/sigc5561_ptmc.htm
in PCI and PCIe formats.
Interesting. How does it work in detail. Does rtpproxy use the DSP card for transcoding, or is rtpproxy bypassed completely and Kamailio talkes with the DSP card directly which works like rtpproxy itself (as I saw the DPS card has a dedicated network inteface)?
rtpproxy interfaces with the card. rtpproxy uses the card's GbE for IP addr's instead of the server. So the idea (more or less) is the server handles signaling / SIP and card handles data / RTP. We keep Kamailio away from the card, so it doesn't have to be in the same server.
So, rtpproxy is more or less only used to control the RTP-forwarder-and-transcoder on the DSP card?
klaus
-
Jeff Brower -
Klaus Darilion -
Raúl Alexis Betancor Santana