What cfg files changes do I need to make to get Kamailio to be a signally only server, yet manipulate the SDP part of the INVITE message to allow remote parties to send media to each other? In Freeswitch terms "bypassmedia". KD
Ironically, nothing. Kamailio doesn't touch the respective parties' SDP unless you invoke an RTP relay (or something else like fix_nated_sdp()).
On May 9, 2018 1:03:19 PM EDT, KamDev Essa kamdevessa@yahoo.com wrote:
What cfg files changes do I need to make to get Kamailio to be a signally only server, yet manipulate the SDP part of the INVITE message to allow remote parties to send media to each other? In Freeswitch terms "bypassmedia". KD
-- Alex
-- Sent via mobile, please forgive typos and brevity.
So all calls that kamailio processes using the default cfg file anchor RTP on the kamailio server? Is it a best architecture to farm out RTP to Freeswitch? Or is the Kamailio RTP proxy a better gig? KD On Wednesday, May 9, 2018, 1:38:04 PM EDT, Alex Balashov abalashov@evaristesys.com wrote:
Ironically, nothing. Kamailio doesn't touch the respective parties' SDP unless you invoke an RTP relay (or something else like fix_nated_sdp()).
On May 9, 2018 1:03:19 PM EDT, KamDev Essa kamdevessa@yahoo.com wrote:
What cfg files changes do I need to make to get Kamailio to be a signally only server, yet manipulate the SDP part of the INVITE message to allow remote parties to send media to each other? In Freeswitch terms "bypassmedia". KD
-- Alex
-- Sent via mobile, please forgive typos and brevity.
_______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
That depends. Start with a more basic question: why do you need RTP relay in the middle at all?
On May 9, 2018 1:54:55 PM EDT, KamDev Essa kamdevessa@yahoo.com wrote:
So all calls that kamailio processes using the default cfg file anchor RTP on the kamailio server? Is it a best architecture to farm out RTP to Freeswitch? Or is the Kamailio RTP proxy a better gig? KD On Wednesday, May 9, 2018, 1:38:04 PM EDT, Alex Balashov abalashov@evaristesys.com wrote:
Ironically, nothing. Kamailio doesn't touch the respective parties' SDP unless you invoke an RTP relay (or something else like fix_nated_sdp()).
On May 9, 2018 1:03:19 PM EDT, KamDev Essa kamdevessa@yahoo.com wrote:
What cfg files changes do I need to make to get Kamailio to be a signally only server, yet manipulate the SDP part of the INVITE
message
to allow remote parties to send media to each other? In Freeswitch terms "bypassmedia". KD
-- Alex
-- Sent via mobile, please forgive typos and brevity.
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Alex
-- Sent via mobile, please forgive typos and brevity.
I dont. I want the 2 end points to talk to each other because I am on AWS with shaky bandwidth stats. It can handle signalingbut not RTP. However the cfg entry modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") gives me to believe (not tcdumped RTP ports yet to prove it ) that Kam anchors RTP @ port 7722. KD On Wednesday, May 9, 2018, 2:02:14 PM EDT, Alex Balashov abalashov@evaristesys.com wrote:
That depends. Start with a more basic question: why do you need RTP relay in the middle at all?
On May 9, 2018 1:54:55 PM EDT, KamDev Essa kamdevessa@yahoo.com wrote:
So all calls that kamailio processes using the default cfg file anchor RTP on the kamailio server? Is it a best architecture to farm out RTP to Freeswitch? Or is the Kamailio RTP proxy a better gig? KD On Wednesday, May 9, 2018, 1:38:04 PM EDT, Alex Balashov abalashov@evaristesys.com wrote:
Ironically, nothing. Kamailio doesn't touch the respective parties' SDP unless you invoke an RTP relay (or something else like fix_nated_sdp()).
On May 9, 2018 1:03:19 PM EDT, KamDev Essa kamdevessa@yahoo.com wrote:
What cfg files changes do I need to make to get Kamailio to be a signally only server, yet manipulate the SDP part of the INVITE
message
to allow remote parties to send media to each other? In Freeswitch terms "bypassmedia". KD
-- Alex
-- Sent via mobile, please forgive typos and brevity.
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Alex
-- Sent via mobile, please forgive typos and brevity.
Correct. So remove all semblance of any RTP proxy and the resulting behaviour will be exactly what you expect.
On May 9, 2018 2:13:22 PM EDT, KamDev Essa kamdevessa@yahoo.com wrote:
I dont. I want the 2 end points to talk to each other because I am on AWS with shaky bandwidth stats. It can handle signalingbut not RTP. However the cfg entry modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") gives me to believe (not tcdumped RTP ports yet to prove it ) that Kam anchors RTP @ port 7722. KD On Wednesday, May 9, 2018, 2:02:14 PM EDT, Alex Balashov abalashov@evaristesys.com wrote:
That depends. Start with a more basic question: why do you need RTP relay in the middle at all?
On May 9, 2018 1:54:55 PM EDT, KamDev Essa kamdevessa@yahoo.com wrote:
So all calls that kamailio processes using the default cfg file anchor RTP on the kamailio server? Is it a best architecture to farm out RTP to Freeswitch? Or is the Kamailio RTP proxy a better gig? KD On Wednesday, May 9, 2018, 1:38:04 PM EDT, Alex Balashov abalashov@evaristesys.com wrote:
Ironically, nothing. Kamailio doesn't touch the respective parties'
SDP
unless you invoke an RTP relay (or something else like fix_nated_sdp()).
On May 9, 2018 1:03:19 PM EDT, KamDev Essa kamdevessa@yahoo.com wrote:
What cfg files changes do I need to make to get Kamailio to be a signally only server, yet manipulate the SDP part of the INVITE
message
to allow remote parties to send media to each other? In Freeswitch terms "bypassmedia". KD
-- Alex
-- Sent via mobile, please forgive typos and brevity.
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Alex
-- Sent via mobile, please forgive typos and brevity.
-- Alex
-- Sent via mobile, please forgive typos and brevity.
If I comment out the modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") as suggested, I do get RTP flowing just in one direction. From the UA to kam and out to the carrier. I get nothing geting back to the end point. In other words 1 way RTP. Looks like its Kam RTPProxy or bust. KD On Wednesday, May 9, 2018, 2:38:08 PM EDT, Alex Balashov abalashov@evaristesys.com wrote:
Correct. So remove all semblance of any RTP proxy and the resulting behaviour will be exactly what you expect.
On May 9, 2018 2:13:22 PM EDT, KamDev Essa kamdevessa@yahoo.com wrote:
I dont. I want the 2 end points to talk to each other because I am on AWS with shaky bandwidth stats. It can handle signalingbut not RTP. However the cfg entry modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") gives me to believe (not tcdumped RTP ports yet to prove it ) that Kam anchors RTP @ port 7722. KD On Wednesday, May 9, 2018, 2:02:14 PM EDT, Alex Balashov abalashov@evaristesys.com wrote:
That depends. Start with a more basic question: why do you need RTP relay in the middle at all?
On May 9, 2018 1:54:55 PM EDT, KamDev Essa kamdevessa@yahoo.com wrote:
So all calls that kamailio processes using the default cfg file anchor RTP on the kamailio server? Is it a best architecture to farm out RTP to Freeswitch? Or is the Kamailio RTP proxy a better gig? KD On Wednesday, May 9, 2018, 1:38:04 PM EDT, Alex Balashov abalashov@evaristesys.com wrote:
Ironically, nothing. Kamailio doesn't touch the respective parties'
SDP
unless you invoke an RTP relay (or something else like fix_nated_sdp()).
On May 9, 2018 1:03:19 PM EDT, KamDev Essa kamdevessa@yahoo.com wrote:
What cfg files changes do I need to make to get Kamailio to be a signally only server, yet manipulate the SDP part of the INVITE
message
to allow remote parties to send media to each other? In Freeswitch terms "bypassmedia". KD
-- Alex
-- Sent via mobile, please forgive typos and brevity.
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Alex
-- Sent via mobile, please forgive typos and brevity.
-- Alex
-- Sent via mobile, please forgive typos and brevity.
You'll want to not load the rtpproxy module, and lose any rtpproxy_*() calls in the actual script.
On May 9, 2018 3:26:43 PM EDT, KamDev Essa kamdevessa@yahoo.com wrote:
If I comment out the modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") as suggested, I do get RTP flowing just in one direction. From the UA to kam and out to the carrier. I get nothing geting back to the end point. In other words 1 way RTP. Looks like its Kam RTPProxy or bust. KD On Wednesday, May 9, 2018, 2:38:08 PM EDT, Alex Balashov abalashov@evaristesys.com wrote:
Correct. So remove all semblance of any RTP proxy and the resulting behaviour will be exactly what you expect.
On May 9, 2018 2:13:22 PM EDT, KamDev Essa kamdevessa@yahoo.com wrote:
I dont. I want the 2 end points to talk to each other because I am on AWS with shaky bandwidth stats. It can handle signalingbut not RTP. However the cfg entry modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") gives me to believe (not tcdumped RTP ports yet to prove it ) that Kam anchors RTP @ port 7722. KD On Wednesday, May 9, 2018, 2:02:14 PM EDT, Alex Balashov abalashov@evaristesys.com wrote:
That depends. Start with a more basic question: why do you need RTP relay in the middle at all?
On May 9, 2018 1:54:55 PM EDT, KamDev Essa kamdevessa@yahoo.com wrote:
So all calls that kamailio processes using the default cfg file
anchor
RTP on the kamailio server? Is it a best architecture to farm out RTP to Freeswitch? Or is the Kamailio RTP proxy a better gig? KD On Wednesday, May 9, 2018, 1:38:04 PM EDT, Alex Balashov abalashov@evaristesys.com wrote:
Ironically, nothing. Kamailio doesn't touch the respective parties'
SDP
unless you invoke an RTP relay (or something else like fix_nated_sdp()).
On May 9, 2018 1:03:19 PM EDT, KamDev Essa kamdevessa@yahoo.com wrote:
What cfg files changes do I need to make to get Kamailio to be a signally only server, yet manipulate the SDP part of the INVITE
message
to allow remote parties to send media to each other? In Freeswitch terms "bypassmedia". KD
-- Alex
-- Sent via mobile, please forgive typos and brevity.
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Alex
-- Sent via mobile, please forgive typos and brevity.
-- Alex
-- Sent via mobile, please forgive typos and brevity.
-- Alex
-- Sent via mobile, please forgive typos and brevity.
That did not help. sip capture shows NATed SDP being sent to carrier. Obviously carrier will not be able to make head nor tails about it. And carrier SDP sent to UA is perfect as its not NATed. So UA ===> Carrier is fine. Carrier ====> UA is no RTP. Does not look like Kam alone can do RTP relase between NATed UA and the outside network entity. I will try Freeswitch piggy back and then let Freeswitch bypass media. Logic in Freeswitch may do the trick. \ KD On Wednesday, May 9, 2018, 3:30:26 PM EDT, Alex Balashov abalashov@evaristesys.com wrote:
You'll want to not load the rtpproxy module, and lose any rtpproxy_*() calls in the actual script.
On May 9, 2018 3:26:43 PM EDT, KamDev Essa kamdevessa@yahoo.com wrote:
If I comment out the modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") as suggested, I do get RTP flowing just in one direction. From the UA to kam and out to the carrier. I get nothing geting back to the end point. In other words 1 way RTP. Looks like its Kam RTPProxy or bust. KD On Wednesday, May 9, 2018, 2:38:08 PM EDT, Alex Balashov abalashov@evaristesys.com wrote:
Correct. So remove all semblance of any RTP proxy and the resulting behaviour will be exactly what you expect.
On May 9, 2018 2:13:22 PM EDT, KamDev Essa kamdevessa@yahoo.com wrote:
I dont. I want the 2 end points to talk to each other because I am on AWS with shaky bandwidth stats. It can handle signalingbut not RTP. However the cfg entry modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") gives me to believe (not tcdumped RTP ports yet to prove it ) that Kam anchors RTP @ port 7722. KD On Wednesday, May 9, 2018, 2:02:14 PM EDT, Alex Balashov abalashov@evaristesys.com wrote:
That depends. Start with a more basic question: why do you need RTP relay in the middle at all?
On May 9, 2018 1:54:55 PM EDT, KamDev Essa kamdevessa@yahoo.com wrote:
So all calls that kamailio processes using the default cfg file
anchor
RTP on the kamailio server? Is it a best architecture to farm out RTP to Freeswitch? Or is the Kamailio RTP proxy a better gig? KD On Wednesday, May 9, 2018, 1:38:04 PM EDT, Alex Balashov abalashov@evaristesys.com wrote:
Ironically, nothing. Kamailio doesn't touch the respective parties'
SDP
unless you invoke an RTP relay (or something else like fix_nated_sdp()).
On May 9, 2018 1:03:19 PM EDT, KamDev Essa kamdevessa@yahoo.com wrote:
What cfg files changes do I need to make to get Kamailio to be a signally only server, yet manipulate the SDP part of the INVITE
message
to allow remote parties to send media to each other? In Freeswitch terms "bypassmedia". KD
-- Alex
-- Sent via mobile, please forgive typos and brevity.
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Alex
-- Sent via mobile, please forgive typos and brevity.
-- Alex
-- Sent via mobile, please forgive typos and brevity.
-- Alex
-- Sent via mobile, please forgive typos and brevity.
_______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Oh, the UAs are NAT'd? Yeah, you're going to need something clever in the middle that can do the RTP latching, then.
-- Alex
-- Sent via mobile, please forgive typos and brevity.
Well then why does Inbound (from carrier to NATed UA) call work. Kam is doing something clever there. Why not when sending the call out. KD On Wednesday, May 9, 2018, 4:34:55 PM EDT, Alex Balashov abalashov@evaristesys.com wrote:
Oh, the UAs are NAT'd? Yeah, you're going to need something clever in the middle that can do the RTP latching, then.
-- Alex
-- Sent via mobile, please forgive typos and brevity.
Re-read the piece of the article related to "RTP latching":
http://blog.csrpswitch.com/server-side-nat-traversal-with-kamailio-the-defin...
In order for RTP to reach a NAT'd endpoint, all other things being equal, the other party has to do RTP latching. This is true of both inbound and outbound calls. Failure to do it will result in one-way audio where the NAT'd party can transmit but cannot receive.
As far as identifying who is doing that latching, that's not clear from your description. Some carrier media gateways do it, in which case you can get away with not having an RTP relay in the middle. Most carrier media gateways are designed for wholesale-ish peering and don't, as a matter of policy, but some do. Or you could have some near-end solutions at work of which you're not aware.
It's also not clear whether you've definitively removed all rtpproxy invocation from your Kamailio config, in both directions. But if you haven't, I'd start there, and given your statement that you don't really want to be in series to the RTP path, figure out what exactly doesn't work. Then add rtpproxy as necessary. Beware the distinction between inbound and outbound call flows.
-- Alex
On Wed, May 09, 2018 at 09:15:48PM +0000, KamDev Essa wrote:
Well then why does Inbound (from carrier to NATed UA) call work. Kam is doing something clever there. Why not when sending the call out. KD On Wednesday, May 9, 2018, 4:34:55 PM EDT, Alex Balashov abalashov@evaristesys.com wrote:
Oh, the UAs are NAT'd? Yeah, you're going to need something clever in the middle that can do the RTP latching, then.
-- Alex
-- Sent via mobile, please forgive typos and brevity.
Response from the carrier matches your description. Looks like their Inbound carrier is latching but the outbound carrier is not and yes they recommended handling the NAT on my end. That said, whats my options here. Is the native rtpproxy scalable? or is it better to go with a Freeswitch farm to handle media proxying. We are looking @ holding in upwards of 10K UAs on one instance of Kamailio. So whats the best architecture or that ? H On Wednesday, May 9, 2018, 5:53:55 PM EDT, Alex Balashov abalashov@evaristesys.com wrote:
Re-read the piece of the article related to "RTP latching":
http://blog.csrpswitch.com/server-side-nat-traversal-with-kamailio-the-defin...
In order for RTP to reach a NAT'd endpoint, all other things being equal, the other party has to do RTP latching. This is true of both inbound and outbound calls. Failure to do it will result in one-way audio where the NAT'd party can transmit but cannot receive.
As far as identifying who is doing that latching, that's not clear from your description. Some carrier media gateways do it, in which case you can get away with not having an RTP relay in the middle. Most carrier media gateways are designed for wholesale-ish peering and don't, as a matter of policy, but some do. Or you could have some near-end solutions at work of which you're not aware.
It's also not clear whether you've definitively removed all rtpproxy invocation from your Kamailio config, in both directions. But if you haven't, I'd start there, and given your statement that you don't really want to be in series to the RTP path, figure out what exactly doesn't work. Then add rtpproxy as necessary. Beware the distinction between inbound and outbound call flows.
-- Alex
On Wed, May 09, 2018 at 09:15:48PM +0000, KamDev Essa wrote:
Well then why does Inbound (from carrier to NATed UA) call work. Kam is doing something clever there. Why not when sending the call out. KD On Wednesday, May 9, 2018, 4:34:55 PM EDT, Alex Balashov abalashov@evaristesys.com wrote: Oh, the UAs are NAT'd? Yeah, you're going to need something clever in the middle that can do the RTP latching, then.
-- Alex
-- Sent via mobile, please forgive typos and brevity.
Update : I piggy backed Freeswitch to Kamailio and have resolved the issue. No media touching us now. Pros and Cons for doing that I know. But we have it working. Thanks for the dialog. KD On Wednesday, May 9, 2018, 8:57:49 PM EDT, KamDev Essa kamdevessa@yahoo.com wrote:
Response from the carrier matches your description. Looks like their Inbound carrier is latching but the outbound carrier is not and yes they recommended handling the NAT on my end. That said, whats my options here. Is the native rtpproxy scalable? or is it better to go with a Freeswitch farm to handle media proxying. We are looking @ holding in upwards of 10K UAs on one instance of Kamailio. So whats the best architecture or that ? H On Wednesday, May 9, 2018, 5:53:55 PM EDT, Alex Balashov abalashov@evaristesys.com wrote:
Re-read the piece of the article related to "RTP latching":
http://blog.csrpswitch.com/server-side-nat-traversal-with-kamailio-the-defin...
In order for RTP to reach a NAT'd endpoint, all other things being equal, the other party has to do RTP latching. This is true of both inbound and outbound calls. Failure to do it will result in one-way audio where the NAT'd party can transmit but cannot receive.
As far as identifying who is doing that latching, that's not clear from your description. Some carrier media gateways do it, in which case you can get away with not having an RTP relay in the middle. Most carrier media gateways are designed for wholesale-ish peering and don't, as a matter of policy, but some do. Or you could have some near-end solutions at work of which you're not aware.
It's also not clear whether you've definitively removed all rtpproxy invocation from your Kamailio config, in both directions. But if you haven't, I'd start there, and given your statement that you don't really want to be in series to the RTP path, figure out what exactly doesn't work. Then add rtpproxy as necessary. Beware the distinction between inbound and outbound call flows.
-- Alex
On Wed, May 09, 2018 at 09:15:48PM +0000, KamDev Essa wrote:
Well then why does Inbound (from carrier to NATed UA) call work. Kam is doing something clever there. Why not when sending the call out. KD On Wednesday, May 9, 2018, 4:34:55 PM EDT, Alex Balashov abalashov@evaristesys.com wrote: Oh, the UAs are NAT'd? Yeah, you're going to need something clever in the middle that can do the RTP latching, then.
-- Alex
-- Sent via mobile, please forgive typos and brevity.
Update. I just checked inbound calls using the "rtpproxy clean" cfg. Inbound calls from Carrier to UA work fine with bidirectional RTP. Its the outbound calls from UA to carrier that have a NAT issue. I am so close :).... KD On Wednesday, May 9, 2018, 4:29:39 PM EDT, KamDev Essa kamdevessa@yahoo.com wrote:
That did not help. sip capture shows NATed SDP being sent to carrier. Obviously carrier will not be able to make head nor tails about it. And carrier SDP sent to UA is perfect as its not NATed. So UA ===> Carrier is fine. Carrier ====> UA is no RTP. Does not look like Kam alone can do RTP relase between NATed UA and the outside network entity. I will try Freeswitch piggy back and then let Freeswitch bypass media. Logic in Freeswitch may do the trick. \ KD On Wednesday, May 9, 2018, 3:30:26 PM EDT, Alex Balashov abalashov@evaristesys.com wrote:
You'll want to not load the rtpproxy module, and lose any rtpproxy_*() calls in the actual script.
On May 9, 2018 3:26:43 PM EDT, KamDev Essa kamdevessa@yahoo.com wrote:
If I comment out the modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") as suggested, I do get RTP flowing just in one direction. From the UA to kam and out to the carrier. I get nothing geting back to the end point. In other words 1 way RTP. Looks like its Kam RTPProxy or bust. KD On Wednesday, May 9, 2018, 2:38:08 PM EDT, Alex Balashov abalashov@evaristesys.com wrote:
Correct. So remove all semblance of any RTP proxy and the resulting behaviour will be exactly what you expect.
On May 9, 2018 2:13:22 PM EDT, KamDev Essa kamdevessa@yahoo.com wrote:
I dont. I want the 2 end points to talk to each other because I am on AWS with shaky bandwidth stats. It can handle signalingbut not RTP. However the cfg entry modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") gives me to believe (not tcdumped RTP ports yet to prove it ) that Kam anchors RTP @ port 7722. KD On Wednesday, May 9, 2018, 2:02:14 PM EDT, Alex Balashov abalashov@evaristesys.com wrote:
That depends. Start with a more basic question: why do you need RTP relay in the middle at all?
On May 9, 2018 1:54:55 PM EDT, KamDev Essa kamdevessa@yahoo.com wrote:
So all calls that kamailio processes using the default cfg file
anchor
RTP on the kamailio server? Is it a best architecture to farm out RTP to Freeswitch? Or is the Kamailio RTP proxy a better gig? KD On Wednesday, May 9, 2018, 1:38:04 PM EDT, Alex Balashov abalashov@evaristesys.com wrote:
Ironically, nothing. Kamailio doesn't touch the respective parties'
SDP
unless you invoke an RTP relay (or something else like fix_nated_sdp()).
On May 9, 2018 1:03:19 PM EDT, KamDev Essa kamdevessa@yahoo.com wrote:
What cfg files changes do I need to make to get Kamailio to be a signally only server, yet manipulate the SDP part of the INVITE
message
to allow remote parties to send media to each other? In Freeswitch terms "bypassmedia". KD
-- Alex
-- Sent via mobile, please forgive typos and brevity.
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Alex
-- Sent via mobile, please forgive typos and brevity.
-- Alex
-- Sent via mobile, please forgive typos and brevity.
-- Alex
-- Sent via mobile, please forgive typos and brevity.
_______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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Alex Balashov -
KamDev Essa