28 Jan
2013
28 Jan
'13
6:03 p.m.
Hello,
I want to separate signaling and media in our architecture but I'm not entirely sure
if the following will work (note that it is still in a design phase and these kinds of
constructs are still new to me):
User Agent 1-------------- Kamailio -------------- User Agent 2
|
|
|
|
|
|
-------------- Media Server Cluster --------------
The signaling will go through Kamailio but the media stream should pass through one of the
media servers in the cluster for each call.
Is it possible to achieve this by having Kamailio instruct the media server which ports to
open and alter sdp information?
I think that the RTP Proxy module does precisely this, but what I still don't
understand is how the rtp stream is passed between these two agents for each call.
Also, is RTP proxy the best way to do this or are there any other third party media
servers which I could use? I don't need any fancy functionality, just basic media
handling/pass-through.
Regards,
Grant
28 Jan
28 Jan
6:11 p.m.
Hi Grant,
On 01/28/2013 11:03 AM, Grant Bagdasarian wrote:
I think that the RTP Proxy module does precisely this,
but what I
still don’t understand is how the rtp stream is passed between these
two agents for each call.
The Kamailio 'rtpproxy' module goes out to the rtpproxy service[1] and
asks it to engage the call, saying: "Can you please allocate and provide
me with a pair of ports for each end of this call?" The rtpproxy
process answers, and Kamailio then writes the IP and ports provided by
rtpproxy into the SDP of both the requests and replies involved in
setting up the call.
The result is that as far as both UAs are concerned, rtpproxy is their
endpoint, and they will send media to it. Kamailio instructs rtpproxy
to bridge both streams.
When the call is torn down, Kamailio tells rtpproxy to disengage and
deallocate the bridge mapping for those streams.
-- Alex
[1] Which may be running on the same host, or may be running on another
host, in a distributed fashion. Kamailio talks to rtpproxy through
rtpproxy's UDP control socket, and more than one rtpproxy may be used,
both for failover and round-robin load distribution. This is readily
baked into the rtpproxy module.
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
7:14 p.m.
Hello Alex,
Thank you for the explanation. It's clear to me now.
One more thing, I've read that it's not recommended to have media being handled by
virtual servers, is this also the case when using rtpproxy? We are using VMWare for
virtualization.
Regards,
Grant
________________________________________
Van: sr-users-bounces(a)lists.sip-router.org [sr-users-bounces(a)lists.sip-router.org] namens
Alex Balashov [abalashov(a)evaristesys.com]
Verzonden: maandag 28 januari 2013 17:11
Aan: sr-users(a)lists.sip-router.org
Onderwerp: Re: [SR-Users] Need help understanding/separating signaling from media
Hi Grant,
On 01/28/2013 11:03 AM, Grant Bagdasarian wrote:
I think that the RTP Proxy module does precisely this,
but what I
still don’t understand is how the rtp stream is passed between these
two agents for each call.
The Kamailio 'rtpproxy' module goes out to the rtpproxy service[1] and
asks it to engage the call, saying: "Can you please allocate and provide
me with a pair of ports for each end of this call?" The rtpproxy
process answers, and Kamailio then writes the IP and ports provided by
rtpproxy into the SDP of both the requests and replies involved in
setting up the call.
The result is that as far as both UAs are concerned, rtpproxy is their
endpoint, and they will send media to it. Kamailio instructs rtpproxy
to bridge both streams.
When the call is torn down, Kamailio tells rtpproxy to disengage and
deallocate the bridge mapping for those streams.
-- Alex
[1] Which may be running on the same host, or may be running on another
host, in a distributed fashion. Kamailio talks to rtpproxy through
rtpproxy's UDP control socket, and more than one rtpproxy may be used,
both for failover and round-robin load distribution. This is readily
baked into the rtpproxy module.
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
7:18 p.m.
New subject: Need help understanding/separating signaling from media
It is indeed not generally recommended, due to the extremely real-time, delay-sensitive
nature of media and the way VMs can subject a user space process like rtpproxy to
distorted I/O scheduling and timing.
However, this seems to be less of a problem as virtualisation evolves closer and closer
"to the metal", and in any case is unlikely to be an issue for relatively small
numbers of concurrent calls, on a non-oversubscribed hypervisor.
Still, it is not recommended from a "best practical" point of view, definitely
not.
Grant Bagdasarian <GB(a)cm.nl> wrote:
Hello Alex,
Thank you for the explanation. It's clear to me now.
One more thing, I've read that it's not recommended to have media being
handled by virtual servers, is this also the case when using rtpproxy?
We are using VMWare for virtualization.
Regards,
Grant
________________________________________
Van: sr-users-bounces(a)lists.sip-router.org
[sr-users-bounces(a)lists.sip-router.org] namens Alex Balashov
[abalashov(a)evaristesys.com]
Verzonden: maandag 28 januari 2013 17:11
Aan: sr-users(a)lists.sip-router.org
Onderwerp: Re: [SR-Users] Need help understanding/separating signaling
from media
Hi Grant,
On 01/28/2013 11:03 AM, Grant Bagdasarian wrote:
--
Sent from my mobile, and thus lacking in the refinement one might expect from a
fully-fledged keyboard.
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
I think that the RTP Proxy module does precisely
this, but what I
still don’t understand is how the rtp stream is passed between these
two agents for each call.
The Kamailio 'rtpproxy' module goes out to the rtpproxy service[1] and
asks it to engage the call, saying: "Can you please allocate and
provide
me with a pair of ports for each end of this call?" The rtpproxy
process answers, and Kamailio then writes the IP and ports provided by
rtpproxy into the SDP of both the requests and replies involved in
setting up the call.
The result is that as far as both UAs are concerned, rtpproxy is their
endpoint, and they will send media to it. Kamailio instructs rtpproxy
to bridge both streams.
When the call is torn down, Kamailio tells rtpproxy to disengage and
deallocate the bridge mapping for those streams.
-- Alex
[1] Which may be running on the same host, or may be running on another
host, in a distributed fashion. Kamailio talks to rtpproxy through
rtpproxy's UDP control socket, and more than one rtpproxy may be used,
both for failover and round-robin load distribution. This is readily
baked into the rtpproxy module.
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
9:06 p.m.
New subject: Need help understanding/separating signaling from media
Alright, thanks! I'll just use virtuals for our sip proxies and physical servers for
media, like I intended to do.
________________________________________
Van: sr-users-bounces(a)lists.sip-router.org [sr-users-bounces(a)lists.sip-router.org] namens
Alex Balashov [abalashov(a)evaristesys.com]
Verzonden: maandag 28 januari 2013 18:18
Aan: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing
List
Onderwerp: Re: [SR-Users] Need help understanding/separating signaling from media
It is indeed not generally recommended, due to the extremely real-time, delay-sensitive
nature of media and the way VMs can subject a user space process like rtpproxy to
distorted I/O scheduling and timing.
However, this seems to be less of a problem as virtualisation evolves closer and closer
"to the metal", and in any case is unlikely to be an issue for relatively small
numbers of concurrent calls, on a non-oversubscribed hypervisor.
Still, it is not recommended from a "best practical" point of view, definitely
not.
Grant Bagdasarian <GB(a)cm.nl> wrote:
Hello Alex,
Thank you for the explanation. It's clear to me now.
One more thing, I've read that it's not recommended to have media being
handled by virtual servers, is this also the case when using rtpproxy?
We are using VMWare for virtualization.
Regards,
Grant
________________________________________
Van: sr-users-bounces(a)lists.sip-router.org
[sr-users-bounces(a)lists.sip-router.org] namens Alex Balashov
[abalashov(a)evaristesys.com]
Verzonden: maandag 28 januari 2013 17:11
Aan: sr-users(a)lists.sip-router.org
Onderwerp: Re: [SR-Users] Need help understanding/separating signaling
from media
Hi Grant,
On 01/28/2013 11:03 AM, Grant Bagdasarian wrote:
--
Sent from my mobile, and thus lacking in the refinement one might expect from a
fully-fledged keyboard.
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
I think that the RTP Proxy module does precisely
this, but what I
still don’t understand is how the rtp stream is passed between these
two agents for each call.
The Kamailio 'rtpproxy' module goes out to the rtpproxy service[1] and
asks it to engage the call, saying: "Can you please allocate and
provide
me with a pair of ports for each end of this call?" The rtpproxy
process answers, and Kamailio then writes the IP and ports provided by
rtpproxy into the SDP of both the requests and replies involved in
setting up the call.
The result is that as far as both UAs are concerned, rtpproxy is their
endpoint, and they will send media to it. Kamailio instructs rtpproxy
to bridge both streams.
When the call is torn down, Kamailio tells rtpproxy to disengage and
deallocate the bridge mapping for those streams.
-- Alex
[1] Which may be running on the same host, or may be running on another
host, in a distributed fashion. Kamailio talks to rtpproxy through
rtpproxy's UDP control socket, and more than one rtpproxy may be used,
both for failover and round-robin load distribution. This is readily
baked into the rtpproxy module.
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
5 Feb
5 Feb
2:12 p.m.
Hello,
Consider the below image like in my original e-mail.
User Agent 1-------------- Kamailio -------------- User Agent 2
|
|
|
|
|
|
-------------- Media Server Cluster --------------
What if UA1 and UA2 are on different remote networks. Kamailio must have two interfaces
connected to both networks in order to send/receive traffic.
Let's say UA1 is on 72.0.0.1 and UA2 is on 10.0.0.1.
Kamailio would have 72.0.0.2 and 10.0.0.2. But a media server must also have IP addresses
in these ranges in order to receive traffic and in turn bridge them.
Would RTPProxy still work in this scenario?
Also, let's consider UA1 to be one of many SIP Trunk suppliers. Each using a different
network. This would mean each media server must have multiple IP addresses configured to
exchange traffic with the suppliers.
How does RTPProxy work in a scenario like that? When allocating ports for each call, does
it also look at the IP address where the call is coming from?
Some routes must be configured on the media servers for multiple remote networks to
communicate with each other, but I think that's out of the scope of how RTPProxy
handles things.
I hope someone could explain this to me.
Thanks!
Grant
-----Original Message-----
From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org]
On Behalf Of Alex Balashov
Sent: Monday, January 28, 2013 5:11 PM
To: sr-users(a)lists.sip-router.org
Subject: Re: [SR-Users] Need help understanding/separating signaling from media
Hi Grant,
On 01/28/2013 11:03 AM, Grant Bagdasarian wrote:
I think that the RTP Proxy module does precisely this,
but what I
still don't understand is how the rtp stream is passed between these
two agents for each call.
The Kamailio 'rtpproxy' module goes out to the rtpproxy service[1] and asks it to
engage the call, saying: "Can you please allocate and provide me with a pair of ports
for each end of this call?" The rtpproxy process answers, and Kamailio then writes
the IP and ports provided by rtpproxy into the SDP of both the requests and replies
involved in setting up the call.
The result is that as far as both UAs are concerned, rtpproxy is their endpoint, and they
will send media to it. Kamailio instructs rtpproxy to bridge both streams.
When the call is torn down, Kamailio tells rtpproxy to disengage and deallocate the bridge
mapping for those streams.
-- Alex
[1] Which may be running on the same host, or may be running on another host, in a
distributed fashion. Kamailio talks to rtpproxy through rtpproxy's UDP control
socket, and more than one rtpproxy may be used, both for failover and round-robin load
distribution. This is readily baked into the rtpproxy module.
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
18 Feb
18 Feb
10:06 a.m.
Could someone please take a look at my previous question below?
-----Original Message-----
From: Grant Bagdasarian
Sent: Tuesday, February 5, 2013 1:12 PM
To: sr-users(a)lists.sip-router.org
Subject: RE: [SR-Users] Need help understanding/separating signaling from media
Hello,
Consider the below image like in my original e-mail.
User Agent 1-------------- Kamailio -------------- User Agent 2
|
|
|
|
|
|
-------------- Media Server Cluster --------------
What if UA1 and UA2 are on different remote networks. Kamailio must have two interfaces
connected to both networks in order to send/receive traffic.
Let's say UA1 is on 72.0.0.1 and UA2 is on 10.0.0.1.
Kamailio would have 72.0.0.2 and 10.0.0.2. But a media server must also have IP addresses
in these ranges in order to receive traffic and in turn bridge them.
Would RTPProxy still work in this scenario?
Also, let's consider UA1 to be one of many SIP Trunk suppliers. Each using a different
network. This would mean each media server must have multiple IP addresses configured to
exchange traffic with the suppliers.
How does RTPProxy work in a scenario like that? When allocating ports for each call, does
it also look at the IP address where the call is coming from?
Some routes must be configured on the media servers for multiple remote networks to
communicate with each other, but I think that's out of the scope of how RTPProxy
handles things.
I hope someone could explain this to me.
Thanks!
Grant
-----Original Message-----
From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org]
On Behalf Of Alex Balashov
Sent: Monday, January 28, 2013 5:11 PM
To: sr-users(a)lists.sip-router.org
Subject: Re: [SR-Users] Need help understanding/separating signaling from media
Hi Grant,
On 01/28/2013 11:03 AM, Grant Bagdasarian wrote:
I think that the RTP Proxy module does precisely this,
but what I
still don't understand is how the rtp stream is passed between these
two agents for each call.
The Kamailio 'rtpproxy' module goes out to the rtpproxy service[1] and asks it to
engage the call, saying: "Can you please allocate and provide me with a pair of ports
for each end of this call?" The rtpproxy process answers, and Kamailio then writes
the IP and ports provided by rtpproxy into the SDP of both the requests and replies
involved in setting up the call.
The result is that as far as both UAs are concerned, rtpproxy is their endpoint, and they
will send media to it. Kamailio instructs rtpproxy to bridge both streams.
When the call is torn down, Kamailio tells rtpproxy to disengage and deallocate the bridge
mapping for those streams.
-- Alex
[1] Which may be running on the same host, or may be running on another host, in a
distributed fashion. Kamailio talks to rtpproxy through rtpproxy's UDP control
socket, and more than one rtpproxy may be used, both for failover and round-robin load
distribution. This is readily baked into the rtpproxy module.
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
26 Feb
26 Feb
5:24 p.m.
Alex wrote:
The Kamailio 'rtpproxy' module goes out to the rtpproxy service[1] and asks it to
engage the call, saying: "Can you please allocate and provide me with a pair of ports
for each end of this call?" The rtpproxy process answers, and Kamailio then writes
the IP and ports provided by rtpproxy into the SDP of both the requests and replies
involved in setting up the call.
What if both endpoints are on different networks?
Endpoint 1 -------------- Kamailio -------------- Endpoint 2
| | |
| | |
| | |
+-------------------- RTPProxy -----------------+
Please correct me if I'm wrong:
1) Endpoint 1 (10.0.0.1) sends an INVITE containing the offer to Kamailio
(10.0.0.2).
2) Kamailio asks RTPProxy to engage the call and allocate two ports.
3) Kamailio writes the IP and port of the RTPProxy (10.0.0.3:1000) in the SDP of the
100 Trying and replies back to Endpoint 1.
4) Kamailio (20.0.0.2) now sends the INVITE with offer to Endpoint 2 (20.0.0.1), but
with modified SDP (20.0.0.3:1001).
5) .....
6) Endpoint 1 sends its media to 10.0.0.3:1000 and Endpoint 2 sends its media to
20.0.0.3:1001.
7) RTPProxy bridges both ends.
What makes it so difficult for me to understand is that both Kamailio and RTPProxy are
connected to two different networks. A call between endpoint 1 and 2 must be bridged by
Kamailio and RTPProxy.
How does Kamailio know it has to tell RTPProxy to engage a call on network 10.0.0.0 and
20.0.0.0 and allocate two ports, 1 for each network?
Regards,
Grant
-----Original Message-----
From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org]
On Behalf Of Alex Balashov
Sent: Monday, January 28, 2013 5:11 PM
To: sr-users(a)lists.sip-router.org
Subject: Re: [SR-Users] Need help understanding/separating signaling from media
Hi Grant,
On 01/28/2013 11:03 AM, Grant Bagdasarian wrote:
I think that the RTP Proxy module does precisely this,
but what I
still don't understand is how the rtp stream is
passed between these
two agents for each call.
The Kamailio 'rtpproxy' module goes out to the rtpproxy service[1] and asks it to
engage the call, saying: "Can you please allocate and provide me with a pair of ports
for each end of this call?" The rtpproxy process answers, and Kamailio then writes
the IP and ports provided by rtpproxy into the SDP of both the requests and replies
involved in setting up the call.
The result is that as far as both UAs are concerned, rtpproxy is their endpoint, and they
will send media to it. Kamailio instructs rtpproxy to bridge both streams.
When the call is torn down, Kamailio tells rtpproxy to disengage and deallocate the bridge
mapping for those streams.
-- Alex
[1] Which may be running on the same host, or may be running on another host, in a
distributed fashion. Kamailio talks to rtpproxy through rtpproxy's UDP control
socket, and more than one rtpproxy may be used, both for failover and round-robin load
distribution. This is readily baked into the rtpproxy module.
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
_______________________________________________
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
13 Mar
13 Mar
10:57 a.m.
Alex wrote:
The Kamailio 'rtpproxy' module goes out to the rtpproxy service[1] and asks it to
engage the call, saying: "Can you please allocate and provide me with a pair of ports
for each end of this call?" The rtpproxy process answers, and Kamailio then writes
the IP and ports provided by rtpproxy into the SDP of both the requests and replies
involved in setting up the call.
What if both endpoints are on different networks?
Endpoint 1 -------------- Kamailio -------------- Endpoint 2
| | |
| | |
| | |
+-------------------- RTPProxy -----------------+
Please correct me if I'm wrong:
1) Endpoint 1 (10.0.0.1) sends an INVITE containing the offer to Kamailio
(10.0.0.2).
2) Kamailio asks RTPProxy to engage the call and allocate two ports.
3) Kamailio writes the IP and port of the RTPProxy (10.0.0.3:1000) in the SDP of the
100 Trying and replies back to Endpoint 1.
4) Kamailio (20.0.0.2) now sends the INVITE with offer to Endpoint 2 (20.0.0.1), but
with modified SDP (20.0.0.3:1001).
5) .....
6) Endpoint 1 sends its media to 10.0.0.3:1000 and Endpoint 2 sends its media to
20.0.0.3:1001.
7) RTPProxy bridges both ends.
What makes it so difficult for me to understand is that both Kamailio and RTPProxy are
connected to two different networks. A call between endpoint 1 and 2 must be bridged by
Kamailio and RTPProxy.
How does Kamailio know it has to tell RTPProxy to engage a call on network 10.0.0.0 and
20.0.0.0 and allocate two ports, 1 for each network?
Regards,
Grant
-----Original Message-----
From:
sr-users-bounces@lists.sip-router.org<mailto:sr-users-bounces@lists.sip-router.org>
[mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Alex Balashov
Sent: Monday, January 28, 2013 5:11 PM
To: sr-users@lists.sip-router.org<mailto:sr-users@lists.sip-router.org>
Subject: Re: [SR-Users] Need help understanding/separating signaling from media
Hi Grant,
On 01/28/2013 11:03 AM, Grant Bagdasarian wrote:
I think that the RTP Proxy module does precisely this,
but what I
still don't understand is how the rtp stream is
passed between these
two agents for each call.
The Kamailio 'rtpproxy' module goes out to the rtpproxy service[1] and asks it to
engage the call, saying: "Can you please allocate and provide me with a pair of ports
for each end of this call?" The rtpproxy process answers, and Kamailio then writes
the IP and ports provided by rtpproxy into the SDP of both the requests and replies
involved in setting up the call.
The result is that as far as both UAs are concerned, rtpproxy is their endpoint, and they
will send media to it. Kamailio instructs rtpproxy to bridge both streams.
When the call is torn down, Kamailio tells rtpproxy to disengage and deallocate the bridge
mapping for those streams.
-- Alex
[1] Which may be running on the same host, or may be running on another host, in a
distributed fashion. Kamailio talks to rtpproxy through rtpproxy's UDP control
socket, and more than one rtpproxy may be used, both for failover and round-robin load
distribution. This is readily baked into the rtpproxy module.
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
_______________________________________________
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
4272
days inactive
4316
days old
8 comments
2 participants
participants (2)
-
Alex Balashov -
Grant Bagdasarian