23 Oct
2009
23 Oct
'09
1:57 a.m.
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
23 Oct
23 Oct
4:15 p.m.
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(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
6:01 p.m.
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(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
6:39 p.m.
Jeff Brower schrieb:
Klaus-
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
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?
9:10 p.m.
Klaus-
Jeff Brower schrieb:
Thanks for your reply. Yes you're right... I think we just need to try it stateless
and measure the performance.
-Jeff
Klaus-
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. 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? 
9:06 p.m.
On Friday 23 October 2009 16:01:41 Jeff Brower wrote:
Klaus-
It's possible to know witch DSP card are you using ?
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. 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 Betancor Santana
Dimensión Virtual
9:32 p.m.
Raúl Alexis-
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
On Friday 23 October 2009 16:01:41 Jeff Brower wrote:
http://www.signalogic.com/sigc5561_ptmc.htm
in PCI and PCIe formats.
Klaus-
It's possible to know witch DSP card are you using ? 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. 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 Betancor Santana
Dimensión Virtual
_______________________________________________
Kamailio (OpenSER) - Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
9:50 p.m.
Jeff Brower schrieb:
Raúl Alexis-
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
On Friday 23 October 2009 16:01:41 Jeff Brower
wrote:
http://www.signalogic.com/sigc5561_ptmc.htm
in PCI and PCIe formats. Klaus-
It's possible to know witch DSP card are you using ? 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.
10:06 p.m.
Klaus-
Jeff Brower schrieb:
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
Raúl Alexis-
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)? On Friday 23 October 2009 16:01:41 Jeff Brower
wrote:
http://www.signalogic.com/sigc5561_ptmc.htm
in PCI and PCIe formats. 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 ?
10:09 p.m.
Jeff Brower schrieb:
Klaus-
So, rtpproxy is more or less only used to control the
RTP-forwarder-and-transcoder on the DSP card?
klaus
Jeff Brower schrieb:
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. Raúl Alexis-
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)? 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.
5509
days inactive
5510
days old
9 comments
3 participants
participants (3)
-
Jeff Brower -
Klaus Darilion -
Raúl Alexis Betancor Santana