[SR-Users] Music on hold with freeswitch as SBC

Thu Dec 22 19:04:47 CET 2011

Thanks Daniel, I wasn't starting rtpproxy properly. So the initial error is
gone. But in my script when the rtpproxy_stream2uac is called i get the
following log:

Dec 22 08:18:44 vps daemon.err /usr/sbin/kamailio[7901]: ERROR: rtpproxy
[rtpproxy.c:1581]: script error -no valid set selected
Dec 22 08:18:44 vps daemon.err /usr/sbin/kamailio[7901]: ERROR: rtpproxy
[rtpproxy_stream.c:113]: no available proxies

I have given the following commands at the beginning of the call:
set_rtp_proxy_set("0");
rtpproxy_manage();

And when hold is pressed I've called stream2uac. Where am I going wrong?
Could you also tell if rtpproxy_stream2uac will be able to play .wav files
directly?

Thanks for your help.

Gautam

On Thu, Dec 22, 2011 at 7:38 AM, Daniel-Constantin Mierla <miconda at gmail.com
> wrote:

>  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>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> 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>
>>>>>  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
>>>> * Cell phone +46 70 593 68 51 <%2B46%2070%20593%2068%2051>, Office +46
>>>> 8 96 40 20 <%2B46%208%2096%2040%2020>, Sweden
>>>>
>>>>
>>>>
>>>>
>>>>  _______________________________________________
>>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>>> sr-users at lists.sip-router.org
>>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>
>>> --
>>> Daniel-Constantin Mierla -- http://www.asipto.com
>>> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>>>
>>>
>>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users at lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
> --
> Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda
>
>
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