Hi,
I don't know about Kamailio or other servers.
But to test simultaneous calls, you don't require different PC. Only you
require is different user ids and hard ware units (if you are testing media
also).
We can same port for entire sip signalling.
For media, we should use different ports for
Hi Vineet,
Actually if you want to know how many concurrent calls any SIP server can
handle then SIPp is a great tool. No matter how SIPp makes calls
(*parallel*/sequentially)
the calls have a duration defined in your script. and SIPp keeps on sending
calls so they all are active for kamailio.
So
Hi,
I want to test kamailio or any other server for it's simultaneous call
rates, form which I mean how much calls it can support simultaneously??
I know about SIPp, but that won't fulfill my purpose, because SIPp doesn't
offer simultaneity; rather it issues calls serially.
BTW, is it possible t
Unfortunately, I've had no luck with the rtpproxy module.
I have rtpproxy server running and support is enabled for it (verified by
logs), however it doesn't seem to record any sessions.
This is my rtpproxy running:
/usr/sbin/rtpproxy -s udp:127.0.0.1 7722 -p /var/run/rtpproxy/rtpproxy.pid
-R -a
Hello,
for curiosity I did a test with r-uri (from the first email):
sip:a...@cl1.oms.ser.v0.pftest.net;play=http://1xx.1xx.1xx.1xx:8080/errfile/err.wav;repeat=forever
and one with r-uri (as suggested by Andreas):
sip:a...@cl1.oms.ser.v0.pftest.net;play=http%3A%2F%2F1xx.1xx.1xx.1xx%3A8080%2Fer
On 06/04/2012 07:29 PM, Iñaki Baz Castillo wrote:
>> My question now is if there are any concerns on your side with handling
>> single-quoted URI params the way we handle double-quoted ones? If we
>> were strict, we'd actually need to remove the handling for double-quoted
>> ones, but for backwards
You can use the rtpproxy module along with the rtpproxy server:
http://kamailio.org/docs/modules/stable/modules/rtpproxy.html#id2529838
http://www.rtpproxy.org/
Regards,
Ovidiu Sas
--
VoIP Embedded, Inc.
http://www.voipembedded.com
On Mon, Jun 4, 2012 at 1:47 PM, Saul Waizer wrote:
> Hello Li
Hello List,
I am looking for a way to record the audio and video of a call. I am
using Kamailio (latest) on Ubuntu.
Does anyone know if there's any existing modules that allow for recording
or any other solution that can provide recording of calls on Kamailio?
Thank you!
2012/6/4 Andreas Granig :
> Now while doing some interop tests with another platform, it turned out
> that double-quotes are not allowed in URI params (see
> https://lists.cs.columbia.edu/pipermail/sip-implementors/2008-May/019335.html,
> and in the grammar the allowed chars in the "unreserved" def
Hi All
Now I am using Kamailio 3.1.5 and RTP proxy 1.1.
Looks both are compatible and working fine.
The RTP Proxy basically sends/receives RTP packets over UDP.
Is there any RTP Proxy available that does send/receive of RTP packets
over TCP and also should be compatible with Kamailio 3.1.5.
If yo
Hi Patrice,
There are couple of characters in the URI which need to be escaped
properly. A correct R-URI would look like this:
sip:a...@cl1.oms.ser.v0.pftest.net;play=http%3A%2F%2F1xx.1xx.1xx.1xx%3A8080%2Ferrfile%2Ferr.wav;repeat=forever
There are string transformations "{s.escape.param}" and
"{
Hi guys,
When I implemented the Path module 6 years ago, I chose to enclose the
received-URI in double-quotes, like this:
Path:
This Path value is stored in usrloc by registrar module, and is
converted to a Route in a subsequent request. This worked fine so far,
because there's the logic in par
Hello,
I'am working on Kamailio version 3.2.2.
I'm doing some test in interaction with Application server and media server.
The applicationServer sends a INVITE sip to Media Server via the Kamailio.
The R-URI has the format
sip:a...@cl1.oms.ser.v0.pftest.net;play=http://1xx.1xx.1xx.1xx:8080/errfil
To make the life easier to spot the hash code in such case, I just
committed a small patch to print it for alg 7:
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=83a558fbad700198a8667ef01f756269d7e2c310
Should be easy to backport if other version is used.
Cheers,
Daniel
On
It's hard to say, most likely not. You find the used hash-function here:
841 unsigned int ds_get_hash(str *x, str *y)
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=modules_k/dispatcher/dispatch.c;h=ae0c51b122d34e29b3474957a64b24ff492cfcd5;hb=HEAD
It was originally implemente
Thanks Sir,
Thats what I expected. Do you think this will change if I use only two
values with huge difference in each other, lets say {26000,29000} ?
Regards,
Sammy
On Mon, Jun 4, 2012 at 3:05 PM, Carsten Bock wrote:
> Hi,
>
> this functionality uses a generic "hash" function to generate an
>
Hi,
this functionality uses a generic "hash" function to generate an
Integer-Value of the PVAR and then reduces this to the number hosts in
your destination set.
If you have only using two different values of the PVAR and only two
hosts in the dispatcher list, it can likely happen, that the same h
Hi,
No, nothing at all. I haven't went too deep into debug logs but no internal
error was appearing. Everything was as calm as ever and I only get one
destination uri.
So maybe what I'm getting from your email is that , for this to work the
way I expect is to have more unique and random values in
Thanks,
The reason why I didn't scroll down to it was because it was not listed
under the functions list. Thanks for this. I'll give it a try.
Next question is, will the values stored in this dialog_vars be matched if
I execute the function *is_in_profile("my-profile") ?*
The problem with the alt
Hello,
to store in a value in a dialog_var record, you have to set a PV for it:
http://kamailio.org/docs/modules/stable/modules_k/dialog.html#id2555403
Alternative, you can use hashtable with a key composed from dialog id
(or callid) and some other string to identify your value.
Cheers,
Dani
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