With the help of members from this mailing list (many thanks!), I finally got Asterisk fronted by Kamailio for LB and REGISTERs and I am able to make a call using the setup that looks like this:
[Kamailio 4.4.2]<->[Asterisk 13.7.2]
Kamailio manages REGISTERs, but also forwarding them to Asterisk.
I am able to make a call, but I get only one way audio or no audio depending on which client made the call (SipDroid->Zoiper I hear one way audio on Zoiper, but no audio if the call is made the other way). I noticed that Kamailio forced direct media between the endpoints in this situation, but my application really needs Asterisk to handle it.
How do I do this? Should I start by forwarding INVITEs to Asterisk? How do I do that?
Any help is appreciated.
Thanks!
Hi Again,
You need to enable NAT handling in your Kamailio (#!define WITH_NAT), then depending upon how your clients will interact with asterisk you may or may not need a media proxy, like RTPproxy. If asterisks can send/receive media directly from the internet then its ok for now, else you definitely need to have rtpproxy/rtpengine in there.
Regards, Sammy
On Tue, Jul 26, 2016 at 10:29 PM, Tickling Contest < tickling.contest@gmail.com> wrote:
With the help of members from this mailing list (many thanks!), I finally got Asterisk fronted by Kamailio for LB and REGISTERs and I am able to make a call using the setup that looks like this:
[Kamailio 4.4.2]<->[Asterisk 13.7.2]
Kamailio manages REGISTERs, but also forwarding them to Asterisk.
I am able to make a call, but I get only one way audio or no audio depending on which client made the call (SipDroid->Zoiper I hear one way audio on Zoiper, but no audio if the call is made the other way). I noticed that Kamailio forced direct media between the endpoints in this situation, but my application really needs Asterisk to handle it.
How do I do this? Should I start by forwarding INVITEs to Asterisk? How do I do that?
Any help is appreciated.
Thanks!
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
Hi Tickling,
Could you please share with us your working config?
With kind regards,
Jurijs
On Wed, Jul 27, 2016 at 8:54 AM, SamyGo govoiper@gmail.com wrote:
Hi Again,
You need to enable NAT handling in your Kamailio (#!define WITH_NAT), then depending upon how your clients will interact with asterisk you may or may not need a media proxy, like RTPproxy. If asterisks can send/receive media directly from the internet then its ok for now, else you definitely need to have rtpproxy/rtpengine in there.
Regards, Sammy
On Tue, Jul 26, 2016 at 10:29 PM, Tickling Contest < tickling.contest@gmail.com> wrote:
With the help of members from this mailing list (many thanks!), I finally got Asterisk fronted by Kamailio for LB and REGISTERs and I am able to make a call using the setup that looks like this:
[Kamailio 4.4.2]<->[Asterisk 13.7.2]
Kamailio manages REGISTERs, but also forwarding them to Asterisk.
I am able to make a call, but I get only one way audio or no audio depending on which client made the call (SipDroid->Zoiper I hear one way audio on Zoiper, but no audio if the call is made the other way). I noticed that Kamailio forced direct media between the endpoints in this situation, but my application really needs Asterisk to handle it.
How do I do this? Should I start by forwarding INVITEs to Asterisk? How do I do that?
Any help is appreciated.
Thanks!
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 Wed, Jul 27, 2016 at 01:54:07AM -0400, SamyGo wrote:
You need to enable NAT handling in your Kamailio (#!define WITH_NAT), then depending upon how your clients will interact with asterisk you may or may not need a media proxy, like RTPproxy. If asterisks can send/receive media directly from the internet then its ok for now, else you definitely need to have rtpproxy/rtpengine in there.
I'd suggest to use rtpengine for all calls, it fixes most problems and uses nearly no resources (with the kernel plugin)
I added the #!define WITH_NAT option, and now the call can only be made one way. RTPProxy was started like so:
$ rtpproxy -l 192.168.1.101 -s udp:localhost:7722 -u kamailio
root@kamailioA:~# netstat -pln | egrep "kamailio|rtpproxy" tcp 0 0 192.168.1.101:5060 0.0.0.0:* LISTEN 10112/kamailio tcp 0 0 127.0.0.1:5060 0.0.0.0:* LISTEN 10112/kamailio udp 0 0 192.168.1.101:5060 0.0.0.0:* 10081/kamailio udp 0 0 127.0.0.1:5060 0.0.0.0:* 10081/kamailio udp 0 0 127.0.0.1:7722 0.0.0.0:* 10042/rtpproxy raw 0 0 0.0.0.0:255 0.0.0.0:* 7 10081/kamailio unix 2 [ ACC ] STREAM LISTENING 33357 10102/kamailio /var/run/kamailio//kamailio_ctl
My full config is at https://gist.github.com/ticklingcontest/e315972c80c82f6dfa23920c7725d60b
BTW, my entire setup, kamailio, asterisk and the phones etc. are in one private network. I think setting realtime endpoint with "direct_media=no" is pointless as all of these interactions are fronted by Kamailio.
What's going on here? Any help is appreciated.
On Wed, Jul 27, 2016 at 10:15 AM, Daniel Tryba d.tryba@pocos.nl wrote:
On Wed, Jul 27, 2016 at 01:54:07AM -0400, SamyGo wrote:
You need to enable NAT handling in your Kamailio (#!define WITH_NAT),
then
depending upon how your clients will interact with asterisk you may or
may
not need a media proxy, like RTPproxy. If asterisks can send/receive
media
directly from the internet then its ok for now, else you definitely need
to
have rtpproxy/rtpengine in there.
I'd suggest to use rtpengine for all calls, it fixes most problems and uses nearly no resources (with the kernel plugin)
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
Hello,
I didn't follow your previous tread, but I suppose you use Kamailio as 'frontdoors' gate and route all your calls to external network using Kamailio. In that way you'd better use RTPEngine ( https://github.com/sipwise/rtpengine) installed on Kamailio machine running with external/internal interfaces.
Easiest thing is to put inside LOCATION route something like that
if route(FROMASTERISK) { rtpengine_manage(force trust-address direction=internal direction=external); } else { rtpengine_manage(force trust-address direction=external direction=internal); }
(in route[FROMASTERISK] put a check to be sure call is comeing from your asterisk)
and also - yes, define WITH_NAT if you're using standart configuration
2016-07-27 22:40 GMT+03:00 Tickling Contest tickling.contest@gmail.com:
I added the #!define WITH_NAT option, and now the call can only be made one way. RTPProxy was started like so:
$ rtpproxy -l 192.168.1.101 -s udp:localhost:7722 -u kamailio
root@kamailioA:~# netstat -pln | egrep "kamailio|rtpproxy" tcp 0 0 192.168.1.101:5060 0.0.0.0:* LISTEN 10112/kamailio tcp 0 0 127.0.0.1:5060 0.0.0.0:* LISTEN 10112/kamailio udp 0 0 192.168.1.101:5060 0.0.0.0:* 10081/kamailio udp 0 0 127.0.0.1:5060 0.0.0.0:* 10081/kamailio udp 0 0 127.0.0.1:7722 0.0.0.0:* 10042/rtpproxy raw 0 0 0.0.0.0:255 0.0.0.0:* 7 10081/kamailio unix 2 [ ACC ] STREAM LISTENING 33357 10102/kamailio /var/run/kamailio//kamailio_ctl
My full config is at https://gist.github.com/ticklingcontest/e315972c80c82f6dfa23920c7725d60b
BTW, my entire setup, kamailio, asterisk and the phones etc. are in one private network. I think setting realtime endpoint with "direct_media=no" is pointless as all of these interactions are fronted by Kamailio.
What's going on here? Any help is appreciated.
On Wed, Jul 27, 2016 at 10:15 AM, Daniel Tryba d.tryba@pocos.nl wrote:
On Wed, Jul 27, 2016 at 01:54:07AM -0400, SamyGo wrote:
You need to enable NAT handling in your Kamailio (#!define WITH_NAT),
then
depending upon how your clients will interact with asterisk you may or
may
not need a media proxy, like RTPproxy. If asterisks can send/receive
media
directly from the internet then its ok for now, else you definitely
need to
have rtpproxy/rtpengine in there.
I'd suggest to use rtpengine for all calls, it fixes most problems and uses nearly no resources (with the kernel plugin)
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
-
Alexandru Covalschi -
Daniel Tryba -
Jurijs Ivolga -
SamyGo -
Tickling Contest