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> 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
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
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--
Daniel-Constantin Mierla -- http://www.asipto.com
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