On Wed, Dec 21, 2011 at 4:50 AM,
Daniel-Constantin Mierla
<miconda@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@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@edvina.net
* Cell phone
+46 70 593
68 51, Office
+46 8 96 40
20, Sweden
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