Can you give the output of:
ps auxw | grep -i rtpproxy
That will show if rtpproxy is running and what is its control socket.
Cheers,
Daniel
On 12/21/11 11:25 PM, Gautam Batra wrote:
I'm not able to set up the rtp proxy module. I have entered the following:
loadmodule "rtpproxy.so"
modparam ("rtpproxy", "rtpproxy_sock", "udp:X.Y.Z.W:22222");
Where X.Y.Z.W is the IP address of my machine (same as that of my SIP
server). But the log shows the following errors:
Dec 21 13:11:12 abc daemon.err /usr/sbin/kamailio[9984]: ERROR:
rtpproxy [rtpproxy.c:1503]: can't send command to a RTP proxy
Dec 21 13:11:12 abc daemon.err /usr/sbin/kamailio[9984]: ERROR:
rtpproxy [rtpproxy.c:1538]: proxy <udp:X.Y.Z.W:22222> does not
respond, disable it
Dec 21 13:11:12 abc daemon.warn /usr/sbin/kamailio[9984]: WARNING:
rtpproxy [rtpproxy.c:1395]: can't get version of the RTP proxy
Dec 21 13:11:12 abc daemon.warn /usr/sbin/kamailio[9984]: WARNING:
rtpproxy [rtpproxy.c:1432]: support for RTP proxy <udp:X.Y.Z.W:22222>
has been disabled temporarily
Could anyone tell what I'm doing wrong? I tried to run rtpproxy
separately on the given port before starting kamailio (rtpproxy -s
udp:X.Y.Z.W:22222), but it didn't work.
On Wed, Dec 21, 2011 at 2:36 PM, Gautam Batra <gautambatr...@gmail.com
<mailto:gautambatr...@gmail.com>> wrote:
I am using Freeswitch as an SBC behind Kamailio, and my external
calls are routed via freeswitch. In those calls the music on hold
works as it is handled by freeswitch. Ideally I would like to
somehow redirect when a call is put on hold to the MOH extension.
The other option is by using rtpproxy. I could not find any
documentation on rtpproxy and would really appreciate it if
someone could lead me to it or give me a brief overview on how to
go about using rtpproxy_stream2uac to play music whenever a call
is put on hold.
On Wed, Dec 21, 2011 at 4:50 AM, Daniel-Constantin Mierla
<mico...@gmail.com <mailto:mico...@gmail.com>> wrote:
Hello,
On 12/21/11 7:49 AM, Olle E. Johansson wrote:
20 dec 2011 kl. 22:40 skrev Gautam Batra:
Hi,
Thanks for your replies. Is it possible to play an
audio file in the case of a re-invite directly from
kamailio instead of freeswitch by using
rtpproxy_stream2uac() or something similar?
Kamailioi is still a proxy and from the endpoint point of
view is not involved in the media plane. If you managed to
do that many
endpoints would ignore the packets or see them as a DOS
attack attempt. Other endpoints might just play them.
In later releases of Asterisk, we lock to the IP address
of the peer and would ignore these. Asterisk used to send
music-on-hold
like this before, but we considered it a security issue
and started reinviting to make Asterisk involved in the
call again to play
music on hold. Asterisk can do that, because it's a b2bua
and is an endpoint in the call. Kamailio can't initiate a
reinvite in the
call.
indeed, kamailio cannot initiate re-invites. You can play an
audio file via rtpproxy and rtpproxy_stream2uac() if you use
rtpproxy relaying from the beginning of the call. Otherwise,
use a sip b2bua which does signaling only until you need to
play audio and do re-invites so it gets in media path.
Besides Asterisk or FreeSWITCH, a lightweight b2bua that
probably offers such functionality is sems (sip express media
server) -- I CC-ed Stefan, he can confirm and even give some
leads of how to do it.
Cheers,
Daniel
/O
Gautam
On Mon, Dec 12, 2011 at 4:50 AM, Olle E.
Johansson<o...@edvina.net <mailto:o...@edvina.net>> wrote:
12 dec 2011 kl. 10:45 skrev Daniel-Constantin Mierla:
Hello,
On 12/9/11 9:04 PM, Gautam Batra wrote:
Hello,
I have a kamailio sip proxy server with
freeswitch acting as SBC. I want to redirect
the call to freeswitch when hold is pressed so
that i can play music on hold. I tried this by
using rewritehostport in case of a re-invite,
but the call drops in that case. Could someone
please help me with this?
it is not possible to redirect established calls
(it breaks the RFC3261), you have to route the
call through freeswitch from its start. Perhaps
you can use freeswitch without relaying the media
in first place and when you have on hold, you set
media patch to go through freeswitch.
The only solution is having FreeSwitch send an invite
with replaces to grab the call. The question is how to
get it back.
/O
---
* Olle E Johansson - o...@edvina.net <mailto:o...@edvina.net>
* Cell phone +46 70 593 68 51
<tel:%2B46%2070%20593%2068%2051>, Office +46 8 96 40 20
<tel:%2B46%208%2096%2040%2020>, Sweden
_______________________________________________
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