13 Jul
2015
13 Jul
'15
6:30 p.m.
Hi
Im using an Asterisk cluster behind Kamailio and some services are running
on Asterisk. Traffic always must go to Asterisk so RTPPROXY and RTPNGINE do
the work and send traffic to Asterisk and traffic from Asterisk go to RTP
Proxy
I have been able to get rtpproxy working and later tried RTPengie to be
able to manage SRTP
I have been able to make SRTP To RTP to Asterisk
Im able to call between RTP using rtpproxy as recording is a must and
rtpengine does not record yet.
But im not able to call between SRTP extensions, i understand also SRTP to
RTP would work as im doing with Asterisk (Only the speak SRTP as rtpengine
trasncode)
I have this code on NATMANAGE but i do not know what could be wrong to
handle SRTP
Any idea? Thanks in advance
if(src_ip==ASTERISK1_LAN || src_ip==ASTERISK2_LAN ){
if (sdp_get_line_startswith("$avp(mline)", "m="))
{
if ($avp(mline) =~ "AVP")
{
add_rr_param(";rtp=AVP-kamtoast");
set_rtp_proxy_set("1");
rtpproxy_manage("rwei");
}
if ($avp(mline) =~ "SAVP")
{
add_rr_param(";rtp=SAVP-kamtoast");
rtpengine_manage("trust-address
direction=external replace-origin replace-session-connection ICE=remove RTP
AVP");
}
}
}
else{
if (sdp_get_line_startswith("$avp(mline)", "m="))
{
if ($avp(mline) =~ "AVP")
{
add_rr_param(";rtp=AVP-asttokam");
set_rtp_proxy_set("1");
rtpproxy_manage("rwie");
}
if ($avp(mline) =~ "SAVP")
{
add_rr_param(";rtp=SAVP-asttokam");
rtpengine_manage("trust-address
direction=external replace-origin replace-session-connection ICE=remove ");
}
}
}
If you need any more info let me know.
14 Jul
14 Jul
3:24 p.m.
New subject: Issue handling SRTP and RTP with rtpproxy and rtpengine
Alberto Sagredo-2 wrote
...
I have been able to make SRTP To RTP to Asterisk
But im not able to call between SRTP extensions, i understand also SRTP to
RTP would work as im doing with Asterisk (Only the speak SRTP as rtpengine
trasncode)
If you need any more info let me know.
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@.sip-router
Hi!
If you make SRTP to RTP to Asterisk, you possibly will need vice versa
conversion (when request coming from Asterisk to client with SRTP).
Can you describe the logic of test case: (UA-A (SRTP) --> Kamailio (make
SRTP->RTP) .... etc.
Because your explanation is difficult to understand.
Cheers!
--
View this message in context:
http://sip-router.1086192.n5.nabble.com/Issue-handling-SRTP-and-RTP-with-rt…
Sent from the Users mailing list archive at Nabble.com.
6:01 p.m.
New subject: Issue handling SRTP and RTP with rtpproxy and rtpengine
Thanks Vasily i have changed a little today using a RTPPROXY route.
Thats what i have right now
But its not working as expected
What i try is to detect if i have SAVP from endpoint and translate to RTP
to ASterisk an later RTP from ASterisk translate to SRTP using rtpengine
I had extrange behaviour using rtpproxy that send SRTP to Asterisk and i
have SRTP calls, i though rtpproxy 2.0 could not manage SRTP calls. but it
pass it to Asterisk
Using RTPengine i have tested with rtpproxy_manage as you see and also with
rtpengine.
If i load both start_recording() feature is lost.
On rtpengine (behind NAT) im using it as:
INTERFACES="192.168.0.178 internal/192.168.0.178 external/192.168.0.179
!EXTERN_IP
On NATMANAGE route i call directly
route(RTPPROXY);
Hope this helps
-----
route[RTPPROXY] {
if (is_method("INVITE")){
if(ds_is_from_list(1)){
if (is_ip_rfc1918("$si")) {
xlog("L_INFO", "LLamada desde los
Asterisk_$si -> RTPPROXY\n");
if (sdp_get_line_startswith("$avp(mline)",
"m="))
{
#!ifdef WITH_RTPENGINE
if ($avp(mline) =~ "SAVP")
{
xlog("L_INFO", "Tenemos SRTP ");
xlog("L_INFO", "Llamada entre Extensiones
-> RTPENGINE INTERNAL");
rtpengine_manage("direction=internal
replace-origin replace-session-connection ICE=remove");
return;
}
#!endif
if ($avp(mline) =~ "AVP")
{
xlog("L_INFO", "Tenemos RTP ");
xlog("L_INFO", "Llamada entre Extensiones
-> RTPROXY ");
#!ifdef WITH_RTPPROXY
set_rtp_proxy_set("1");
rtpproxy_manage("fwei");
start_recording();
#!endif
#!ifdef WITH_RTPENGINE
set_rtp_proxy_set("2");
rtpproxy_manage("ie");
#!endif
}
}
}
}else if(!ds_is_from_list()){
if (sdp_get_line_startswith("$avp(mline)",
"m="))
{
#!ifdef WITH_RTPENGINE
if ($avp(mline) =~ "SAVP")
{
xlog("L_INFO", "Tenemos SRTP ");
xlog("L_INFO", "Llamada entre Extensiones
-> RTPENGINE EXTERNAL ");
rtpengine_manage("direction=external
replace-origin replace-session-connection ICE=remove");
return;
}
#!endif
if ($avp(mline) =~ "AVP")
{
xlog("L_INFO", "Tenemos RTP ");
xlog("L_INFO", "Llamada entre Extensiones
-> RTPROXY ");
#!ifdef WITH_RTPPROXY
set_rtp_proxy_set("1");
rtpproxy_manage("fwie");
start_recording();
#!endif
#!ifdef WITH_RTPENGINE
set_rtp_proxy_set("2");
rtpproxy_manage("ei");
#!endif
}
}
}
}
}
2015-07-14 14:24 GMT+02:00 Vasiliy Ganchev <vasiliy.ganchev(a)wildix.com>om>:
Alberto Sagredo-2 wrote
...
I have been able to make SRTP To RTP to Asterisk
But im not able to call between SRTP extensions, i understand also SRTP
to
RTP would work as im doing with Asterisk (Only
the speak SRTP as
rtpengine
trasncode)
If you need any more info let me know.
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@.sip-router
Hi!
If you make SRTP to RTP to Asterisk, you possibly will need vice versa
conversion (when request coming from Asterisk to client with SRTP).
Can you describe the logic of test case: (UA-A (SRTP) --> Kamailio (make
SRTP->RTP) .... etc.
Because your explanation is difficult to understand.
Cheers!
--
View this message in context:
http://sip-router.1086192.n5.nabble.com/Issue-handling-SRTP-and-RTP-with-rt…
Sent from the Users mailing list archive at Nabble.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
6:16 p.m.
New subject: Issue handling SRTP and RTP with rtpproxy and rtpengine
Alberto Sagredo-2 wrote
Thanks Vasily i have changed a little today using a
RTPPROXY route.
Thats what i have right now
But its not working as expected
What i try is to detect if i have SAVP from endpoint and translate to RTP
to ASterisk an later RTP from ASterisk translate to SRTP using rtpengine
I had extrange behaviour using rtpproxy that send SRTP to Asterisk and i
have SRTP calls, i though rtpproxy 2.0 could not manage SRTP calls. but it
pass it to Asterisk
Using RTPengine i have tested with rtpproxy_manage as you see and also
with
rtpengine.
If i load both start_recording() feature is lost.
On rtpengine (behind NAT) im using it as:
INTERFACES="192.168.0.178 internal/192.168.0.178 external/192.168.0.179
!EXTERN_IP
On NATMANAGE route i call directly
route(RTPPROXY);
Hope this helps
-----
route[RTPPROXY] {
if (is_method("INVITE")){
if(ds_is_from_list(1)){
if (is_ip_rfc1918("$si")) {
xlog("L_INFO", "LLamada desde los
Asterisk_$si -> RTPPROXY\n");
if (sdp_get_line_startswith("$avp(mline)",
"m="))
{
#!ifdef WITH_RTPENGINE
if ($avp(mline) =~ "SAVP")
{
xlog("L_INFO", "Tenemos SRTP ");
xlog("L_INFO", "Llamada entre Extensiones
-> RTPENGINE INTERNAL");
rtpengine_manage("direction=internal
replace-origin replace-session-connection ICE=remove");
return;
}
#!endif
if ($avp(mline) =~ "AVP")
{
xlog("L_INFO", "Tenemos RTP ");
xlog("L_INFO", "Llamada entre Extensiones
-> RTPROXY ");
#!ifdef WITH_RTPPROXY
set_rtp_proxy_set("1");
rtpproxy_manage("fwei");
start_recording();
#!endif
#!ifdef WITH_RTPENGINE
set_rtp_proxy_set("2");
rtpproxy_manage("ie");
#!endif
}
}
}
}else if(!ds_is_from_list()){
if (sdp_get_line_startswith("$avp(mline)",
"m="))
{
#!ifdef WITH_RTPENGINE
if ($avp(mline) =~ "SAVP")
{
xlog("L_INFO", "Tenemos SRTP ");
xlog("L_INFO", "Llamada entre Extensiones
-> RTPENGINE EXTERNAL ");
rtpengine_manage("direction=external
replace-origin replace-session-connection ICE=remove");
return;
}
#!endif
if ($avp(mline) =~ "AVP")
{
xlog("L_INFO", "Tenemos RTP ");
xlog("L_INFO", "Llamada entre Extensiones
-> RTPROXY ");
#!ifdef WITH_RTPPROXY
set_rtp_proxy_set("1");
rtpproxy_manage("fwie");
start_recording();
#!endif
#!ifdef WITH_RTPENGINE
set_rtp_proxy_set("2");
rtpproxy_manage("ei");
#!endif
}
}
}
}
}
2015-07-14 14:24 GMT+02:00 Vasiliy Ganchev <
vasiliy.ganchev@
>:
> Alberto Sagredo-2 wrote
> > ...
> > I have been able to make SRTP To RTP to Asterisk
> >
> > But im not able to call between SRTP extensions, i understand also SRTP
> to
> > RTP would work as im doing with Asterisk (Only the speak SRTP as
> rtpengine
> > trasncode)
> >
> >
> > If you need any more info let me know.
> >
> > _______________________________________________
> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>
>
sr-users@.sip-router
>
> >
> Hi!
> If you make SRTP to RTP to Asterisk, you possibly will need vice versa
> conversion (when request coming from Asterisk to client with SRTP).
>
> Can you describe the logic of test case: (UA-A (SRTP) --> Kamailio (make
> SRTP->RTP) .... etc.
>
> Because your explanation is difficult to understand.
>
> Cheers!
>
>
>
> --
> View this message in context:
>
http://sip-router.1086192.n5.nabble.com/Issue-handling-SRTP-and-RTP-with-rt…
> Sent from the Users mailing list archive at Nabble.com.
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@.sip-router
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@.sip-router
What about ICE, where it has to work? (client->Kamailio - yes,
Kamailio->Asterisk - no) or somehow else.
For your description, I think you need to add something like this:
- Kamailio -> Asterisk
rtpengine_manage("...............RTP/AVP"); ///// this will change
profile to RTP/AVP
- Asterisk -> Kamailio
rtpengine_manage("...............RTP/SAVPF"); ///// this will make
backward changes
Also read thoroughly the meaning and usage of "direction" parameter, I think
you have little misunderstanding of how it works (maybe I'm wrong and you
use it as it has to be, but re-read it anyway)
--
View this message in context:
http://sip-router.1086192.n5.nabble.com/Issue-handling-SRTP-and-RTP-with-rt…
Sent from the Users mailing list archive at Nabble.com.
6:26 p.m.
New subject: Issue handling SRTP and RTP with rtpproxy and rtpengine
Thanks Vasily
direction parameter i understood it was as ie options on rtpproxy, maybe im
wrong.
About them , i was ngreping all trace and that was the option to get SDP
correct mapped using RTPengine..
WIll check with your comments and let you know back.
BR
2015-07-14 17:16 GMT+02:00 Vasiliy Ganchev <vasiliy.ganchev(a)wildix.com>om>:
Alberto Sagredo-2 wrote
Thanks Vasily i have changed a little today using
a RTPPROXY route.
Thats what i have right now
But its not working as expected
What i try is to detect if i have SAVP from endpoint and translate to RTP
to ASterisk an later RTP from ASterisk translate to SRTP using rtpengine
I had extrange behaviour using rtpproxy that send SRTP to Asterisk and i
have SRTP calls, i though rtpproxy 2.0 could not manage SRTP calls. but
it
pass it to Asterisk
Using RTPengine i have tested with rtpproxy_manage as you see and also
with
rtpengine.
If i load both start_recording() feature is lost.
On rtpengine (behind NAT) im using it as:
INTERFACES="192.168.0.178 internal/192.168.0.178 external/192.168.0.179
!EXTERN_IP
On NATMANAGE route i call directly
route(RTPPROXY);
Hope this helps
-----
route[RTPPROXY] {
if (is_method("INVITE")){
if(ds_is_from_list(1)){
if (is_ip_rfc1918("$si")) {
xlog("L_INFO", "LLamada desde los
Asterisk_$si -> RTPPROXY\n");
if (sdp_get_line_startswith("$avp(mline)",
"m="))
{
#!ifdef WITH_RTPENGINE
if ($avp(mline) =~ "SAVP")
{
xlog("L_INFO", "Tenemos SRTP ");
xlog("L_INFO", "Llamada entre Extensiones
-> RTPENGINE INTERNAL");
rtpengine_manage("direction=internal
replace-origin replace-session-connection ICE=remove");
return;
}
#!endif
if ($avp(mline) =~ "AVP")
{
xlog("L_INFO", "Tenemos RTP ");
xlog("L_INFO", "Llamada entre Extensiones
-> RTPROXY ");
#!ifdef WITH_RTPPROXY
set_rtp_proxy_set("1");
rtpproxy_manage("fwei");
start_recording();
#!endif
#!ifdef WITH_RTPENGINE
set_rtp_proxy_set("2");
rtpproxy_manage("ie");
#!endif
}
}
}
}else if(!ds_is_from_list()){
if (sdp_get_line_startswith("$avp(mline)",
"m="))
{
#!ifdef WITH_RTPENGINE
if ($avp(mline) =~ "SAVP")
{
xlog("L_INFO", "Tenemos SRTP ");
xlog("L_INFO", "Llamada entre Extensiones
-> RTPENGINE EXTERNAL ");
rtpengine_manage("direction=external
replace-origin replace-session-connection ICE=remove");
return;
}
#!endif
if ($avp(mline) =~ "AVP")
{
xlog("L_INFO", "Tenemos RTP ");
xlog("L_INFO", "Llamada entre Extensiones
-> RTPROXY ");
#!ifdef WITH_RTPPROXY
set_rtp_proxy_set("1");
rtpproxy_manage("fwie");
start_recording();
#!endif
#!ifdef WITH_RTPENGINE
set_rtp_proxy_set("2");
rtpproxy_manage("ei");
#!endif
}
}
}
}
}
2015-07-14 14:24 GMT+02:00 Vasiliy Ganchev <
vasiliy.ganchev@
>:
> Alberto Sagredo-2 wrote
> > ...
> > I have been able to make SRTP To RTP to Asterisk
> >
> > But im not able to call between SRTP extensions, i understand also
SRTP
> to
> > RTP would work as im doing with Asterisk (Only the speak SRTP as
> rtpengine
> > trasncode)
> >
> >
> > If you need any more info let me know.
> >
> > _______________________________________________
> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
>
>
http://sip-router.1086192.n5.nabble.com/Issue-handling-SRTP-and-RTP-with-rt…
sr-users@.sip-router
>
>>
> Hi!
> If you make SRTP to RTP to Asterisk, you possibly will need vice versa
> conversion (when request coming from Asterisk to client with SRTP).
>
> Can you describe the logic of test case: (UA-A (SRTP) --> Kamailio (make
> SRTP->RTP) .... etc.
>
> Because your explanation is difficult to understand.
>
> Cheers!
>
>
>
> --
> View this message in context:
> > Sent from the Users mailing list archive at
Nabble.com.
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>
sr-users@.sip-router
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@.sip-router
What about ICE, where it has to work? (client->Kamailio - yes,
Kamailio->Asterisk - no) or somehow else.
For your description, I think you need to add something like this:
- Kamailio -> Asterisk
rtpengine_manage("...............RTP/AVP"); ///// this will change
profile to RTP/AVP
- Asterisk -> Kamailio
rtpengine_manage("...............RTP/SAVPF"); ///// this will make
backward changes
Also read thoroughly the meaning and usage of "direction" parameter, I
think
you have little misunderstanding of how it works (maybe I'm wrong and you
use it as it has to be, but re-read it anyway)
--
View this message in context:
http://sip-router.1086192.n5.nabble.com/Issue-handling-SRTP-and-RTP-with-rt…
Sent from the Users mailing list archive at Nabble.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
6:33 p.m.
New subject: Issue handling SRTP and RTP with rtpproxy and rtpengine
On Tuesday 14 July 2015 17:01:38 Alberto Sagredo wrote:
What i try is to detect if i have SAVP from endpoint
and translate to RTP
to ASterisk an later RTP from ASterisk translate to SRTP using rtpengine
Why not just handle SRTP with asterisk and record there?
7:19 p.m.
New subject: Issue handling SRTP and RTP with rtpproxy and rtpengine
Hi daniel
In my tests rtpproxy recording waste less resources than asterisk
That was one of the reasons
BR
ALBERTO
El martes, 14 de julio de 2015, Daniel Tryba <d.tryba(a)pocos.nl> escribió:
On Tuesday 14 July 2015 17:01:38 Alberto Sagredo
wrote:
What i try is to detect if i have SAVP from
endpoint and translate to RTP
to ASterisk an later RTP from ASterisk translate to SRTP using rtpengine
Why not just handle SRTP with asterisk and record there?
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org <javascript:;>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
7:46 p.m.
New subject: Issue handling SRTP and RTP with rtpproxy and rtpengine
On Tuesday 14 July 2015 18:19:02 Alberto Sagredo wrote:
In my tests rtpproxy recording waste less resources
than asterisk
That was one of the reasons
How much time have you spend so far on a problem that asterisk can handle out
of the box? ;)
I'd love to do this with kamailio/rtpengine (I don't record), but sofar the
blunt quickfix is to use asterisk. I needed a transcoder anyway and handling
RTP/SRTP conversions when either endpoint needs it is simple.
15 Jul
15 Jul
9:44 a.m.
New subject: Issue handling SRTP and RTP with rtpproxy and rtpengine
Hi Daniel
Kamailio is for hard people and fun :)
Thanks Visily i finnaly got it working with your tip. You were right about
internal external options instead direction=...
Here its some code to someone could need it
route[RTPPROXY] {
if (is_method("INVITE")){
if(ds_is_from_list(1)){
if (is_ip_rfc1918("$si")) {
if (sdp_get_line_startswith("$avp(mline)",
"m="))
{
#!ifdef WITH_RTPENGINE
if ($avp(mline) =~ "SAVP")
{
xlog("L_INFO", "We got SRTP ");
rtpengine_manage("trust-address internal
external replace-origin replace-session-connection ICE=remove ");
return;
}
#!endif
if ($avp(mline) =~ "AVP")
{
xlog("L_INFO", "We got RTP ");
#!ifdef WITH_RTPPROXY
set_rtp_proxy_set("1");
rtpproxy_manage("fwei");
start_recording();
#!endif
#!ifdef WITH_RTPENGINE
rtpengine_manage("trust-address internal
external replace-origin replace-session-connection ICE=remove ");
#!endif
}
}
}
}
else if(!ds_is_from_list()){
if (sdp_get_line_startswith("$avp(mline)",
"m="))
{
#!ifdef WITH_RTPENGINE
if ($avp(mline) =~ "SAVP")
{
xlog("L_INFO", "We got SRTP ");
rtpengine_manage("external internal
replace-origin replace-session-connection ICE=remove RTP AVP");
return;
}
#!endif
if ($avp(mline) =~ "AVP")
{
xlog("L_INFO", "We got RTP ");
#!ifdef WITH_RTPPROXY
set_rtp_proxy_set("1");
rtpproxy_manage("fwie");
start_recording();
#!endif
#!ifdef WITH_RTPENGINE
rtpengine_manage("external internal
replace-origin replace-session-connection ICE=remove RTP AVP");
#!endif
}
}
}
}
}
2015-07-14 18:46 GMT+02:00 Daniel Tryba <d.tryba(a)pocos.nl>nl>:
On Tuesday 14 July 2015 18:19:02 Alberto Sagredo
wrote:
In my tests rtpproxy recording waste less
resources than asterisk
That was one of the reasons
How much time have you spend so far on a problem that asterisk can handle
out
of the box? ;)
I'd love to do this with kamailio/rtpengine (I don't record), but sofar the
blunt quickfix is to use asterisk. I needed a transcoder anyway and
handling
RTP/SRTP conversions when either endpoint needs it is simple.
_______________________________________________
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
4:42 p.m.
New subject: Issue handling SRTP and RTP with rtpproxy and rtpengine
On Wednesday 15 July 2015 08:44:13 Alberto Sagredo wrote:
Kamailio is for hard people and fun :)
No comment on this one....
Thanks Visily i finnaly got it working with your tip.
You were right about
internal external options instead direction=...
Here its some code to someone could need it
Thanks for sharing, I'll see if I can migrate SRTP/RTP conversion to rtpengine
in a distant future.
4:49 p.m.
New subject: Issue handling SRTP and RTP with rtpproxy and rtpengine
You are welcome!
2015-07-15 15:42 GMT+02:00 Daniel Tryba <d.tryba(a)pocos.nl>nl>:
On Wednesday 15 July 2015 08:44:13 Alberto Sagredo
wrote:
Kamailio is for hard people and fun :)
No comment on this one....
Thanks Visily i finnaly got it working with your
tip. You were right
about
internal external options instead direction=...
Here its some code to someone could need it
Thanks for sharing, I'll see if I can migrate SRTP/RTP conversion to
rtpengine
in a distant future.
_______________________________________________
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
5:59 p.m.
New subject: Issue handling SRTP and RTP with rtpproxy and rtpengine
On 07/15/2015 08:44 AM, Alberto Sagredo wrote:
Hi Alberto,
can you also share part of the relevant place where you are calling that route?
Cheers,
Roberto Fichera.
Hi Daniel
Kamailio is for hard people and fun :)
Thanks Visily i finnaly got it working with your tip. You were right about internal
external options instead direction=...
Here its some code to someone could need it
route[RTPPROXY] {
if (is_method("INVITE")){
if(ds_is_from_list(1)){
if (is_ip_rfc1918("$si")) {
if (sdp_get_line_startswith("$avp(mline)",
"m="))
{
#!ifdef WITH_RTPENGINE
if ($avp(mline) =~ "SAVP")
{
xlog("L_INFO", "We got SRTP ");
rtpengine_manage("trust-address internal external
replace-origin
replace-session-connection ICE=remove ");
return;
}
#!endif
if ($avp(mline) =~ "AVP")
{
xlog("L_INFO", "We got RTP ");
#!ifdef WITH_RTPPROXY
set_rtp_proxy_set("1");
rtpproxy_manage("fwei");
start_recording();
#!endif
#!ifdef WITH_RTPENGINE
rtpengine_manage("trust-address internal external
replace-origin
replace-session-connection ICE=remove ");
#!endif
}
}
}
}
else if(!ds_is_from_list()){
if (sdp_get_line_startswith("$avp(mline)",
"m="))
{
#!ifdef WITH_RTPENGINE
if ($avp(mline) =~ "SAVP")
{
xlog("L_INFO", "We got SRTP ");
rtpengine_manage("external internal replace-origin
replace-session-connection
ICE=remove RTP AVP");
return;
}
#!endif
if ($avp(mline) =~ "AVP")
{
xlog("L_INFO", "We got RTP ");
#!ifdef WITH_RTPPROXY
set_rtp_proxy_set("1");
rtpproxy_manage("fwie");
start_recording();
#!endif
#!ifdef WITH_RTPENGINE
rtpengine_manage("external internal replace-origin
replace-session-connection
ICE=remove RTP AVP");
#!endif
}
}
}
}
}
2015-07-14 18:46 GMT+02:00 Daniel Tryba <d.tryba(a)pocos.nl
<mailto:d.tryba@pocos.nl>>:
On Tuesday 14 July 2015 18:19:02 Alberto Sagredo wrote:
In my tests rtpproxy recording waste less
resources than asterisk
That was one of the reasons
How much time have you spend so far on a problem that asterisk can handle out
of the box? ;)
I'd love to do this with kamailio/rtpengine (I don't record), but sofar the
blunt quickfix is to use asterisk. I needed a transcoder anyway and handling
RTP/SRTP conversions when either endpoint needs it is simple.
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7:11 p.m.
New subject: Issue handling SRTP and RTP with rtpproxy and rtpengine
Roberto just call it on NATMANAGE as shown. I route everything now thru
that route..
You have to configure rtpengine to use internal external interfaces
route[NATMANAGE] {
#!ifdef WITH_NAT
# if (is_request()) {
# if(has_totag()) {
# if(check_route_param("nat=yes")) {
# setbflag(FLB_NATB);
# }
# }
# }
# if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))){
# return;
# }
# route(RTPPROXY);
route(RTPPROXY);
if (is_request()) {
if (!has_totag()) {
add_rr_param(";nat=yes");
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
#!endif
return;
}
2015-07-15 16:59 GMT+02:00 Roberto Fichera <kernel(a)tekno-soft.it>it>:
On 07/15/2015 08:44 AM, Alberto Sagredo wrote:
Hi Alberto,
can you also share part of the relevant place where you are calling that
route?
Cheers,
Roberto Fichera.
Hi Daniel
Kamailio is for hard people and fun :)
Thanks Visily i finnaly got it working with your tip. You were right
about internal external options instead direction=...
Here its some code to someone could need it
route[RTPPROXY] {
if (is_method("INVITE")){
if(ds_is_from_list(1)){
if (is_ip_rfc1918("$si")) {
if (sdp_get_line_startswith("$avp(mline)",
"m="))
{
#!ifdef WITH_RTPENGINE
if ($avp(mline) =~ "SAVP")
{
xlog("L_INFO", "We got SRTP ");
rtpengine_manage("trust-address internal
external replace-origin replace-session-connection ICE=remove ");
return;
}
#!endif
if ($avp(mline) =~ "AVP")
{
xlog("L_INFO", "We got RTP ");
#!ifdef WITH_RTPPROXY
set_rtp_proxy_set("1");
rtpproxy_manage("fwei");
start_recording();
#!endif
#!ifdef WITH_RTPENGINE
rtpengine_manage("trust-address internal
external replace-origin replace-session-connection ICE=remove ");
#!endif
}
}
}
}
else if(!ds_is_from_list()){
if (sdp_get_line_startswith("$avp(mline)",
"m="))
{
#!ifdef WITH_RTPENGINE
if ($avp(mline) =~ "SAVP")
{
xlog("L_INFO", "We got SRTP ");
rtpengine_manage("external internal
replace-origin replace-session-connection ICE=remove RTP AVP");
return;
}
#!endif
if ($avp(mline) =~ "AVP")
{
xlog("L_INFO", "We got RTP ");
#!ifdef WITH_RTPPROXY
set_rtp_proxy_set("1");
rtpproxy_manage("fwie");
start_recording();
#!endif
#!ifdef WITH_RTPENGINE
rtpengine_manage("external internal
replace-origin replace-session-connection ICE=remove RTP AVP");
#!endif
}
}
}
}
}
2015-07-14 18:46 GMT+02:00 Daniel Tryba <d.tryba(a)pocos.nl>nl>:
On Tuesday 14 July 2015 18:19:02 Alberto Sagredo
wrote:
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
listsr-users@lists.sip-router.orghttp://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
In my tests rtpproxy recording waste less
resources than asterisk
That was one of the reasons
How much time have you spend so far on a problem that asterisk can handle
out
of the box? ;)
I'd love to do this with kamailio/rtpengine (I don't record), but sofar
the
blunt quickfix is to use asterisk. I needed a transcoder anyway and
handling
RTP/SRTP conversions when either endpoint needs it is simple.
_______________________________________________
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
3421
days inactive
3423
days old
12 comments
4 participants
participants (4)
-
Alberto Sagredo -
Daniel Tryba -
Roberto Fichera -
Vasiliy Ganchev