22 Dec
2011
22 Dec
'11
9:25 a.m.
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(a)gmail.com>wrote;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(a)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,
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
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.
/O
--
Daniel-Constantin Mierla -- http://www.asipto.com
http://linkedin.com/in/miconda -- http://twitter.com/miconda
Gautam
On Mon, Dec 12, 2011 at 4:50 AM, Olle E. Johansson<oej(a)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(a)edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
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