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

Gautam Batra gautambatra24 at gmail.com
Wed Dec 21 20:36:57 CET 2011


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, Office +46 8 96 40 20, Sweden
>>
>>
>>
>>
>> ______________________________**_________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
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>>
>
> --
> Daniel-Constantin Mierla -- http://www.asipto.com
> http://linkedin.com/in/miconda -- http://twitter.com/miconda
>
>
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