[SR-Users] Music on hold with freeswitch as SBC
Daniel-Constantin Mierla
miconda at gmail.com
Thu Dec 22 13:38:42 CET 2011
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 <gautambatra24 at gmail.com
> <mailto:gautambatra24 at 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
> <miconda at gmail.com <mailto:miconda at 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<oej at edvina.net <mailto:oej at 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 - oej at edvina.net <mailto:oej at 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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>
> --
> Daniel-Constantin Mierla -- http://www.asipto.com
> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>
>
>
>
>
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--
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda
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