Hi Daniel,
yes, I hear the audio going both ways ok.
In the meantime I succeeded to record something by rtpproxy
but in best case only one direction, I can hear just the callee,
but not the caller.
In the meantime I have used start_recording in route[NATMANAGE]
but before rtpproxy_manage("co"):
--------------------------------------------------
if (is_method("INVITE") and (status=="200")) {
start_recording();
}
rtpproxy_manage("co");
--------------------------------------------------
After reading of your message I tested also with
start_recording after rtpproxy_manage("co"):
--------------------------------------------------
rtpproxy_manage("co");
if (is_method("INVITE") and (status=="200")) {
start_recording();
}
--------------------------------------------------
But the result was the same:
I have 2 SIP users:
31 (X-Lite on Windows, static address 192.168.0.11)
35 (Android Zoiper, IP address 192.168.0.29 (by DHCP))
Kamailio has static address 192.168.0.13.
When 35 calls 31 then I hear 1 direction in the recorded rtp file,
the voice of callee.
When 31 calls 35 then I don't hear anything in the recorded file.
I don't understand why this difference when user 31 or 35 initiates
a call!?
But I hear audio conversation in both directions, just from the
recorded file in one direction, or even nothing.
I get 2 files per one call: <call-id>.rtp and <call-id>.rtcp
How many files should rtpproxy produce for one call?
Should also a file or two for 2nd channel be generated?
How can I record both audio ways?
Thank you
Marko
The whole function route[NATMANAGE]:
# RTPProxy control
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;
if (is_method("INVITE") and (status=="200")) {
start_recording();
}
rtpproxy_manage("co");
if (is_request()) {
if (!has_totag()) {
if(t_is_branch_route()) {
add_rr_param(";nat=yes");
}
}
}
if (is_reply()) {
if(isbflagset(FLB_NATB)) {
set_contact_alias();
}
}
#!endif
return;
}
----- Original Message -----
From: Daniel-Constantin Mierla <miconda(a)gmail.com>
To: Marko Tirs <marko.tirs(a)yahoo.com>om>; Kamailio (SER) - Users Mailing List
<sr-users(a)lists.sip-router.org>
Sent: Tuesday, March 14, 2017 5:23 PM
Subject: Re: [SR-Users] rtpproxy doesn't record after start_recording
Hello,
I think you can call the start recording function just after
rtpproxy_manage().
If you don't use the start recording function, is the audio going both
ways without problems?
Cheers,
Daniel