> used only when there was a re-invite (like putting the call on hold).
>
> Cheers,
> Daniel
>
> On 04.05.17 17:29, Yufei Tao wrote:
>
> Hi,
>
> Just a quick question: if I use the Kamailio rtpproxy module with the
Sippy
> RTPproxy, can I stream audio within a call
Hi,
Just a quick question: if I use the Kamailio rtpproxy module with the Sippy
RTPproxy, can I stream audio within a call? Or is it only possible before
call audio is established?
For example can I mix/playback audio file into an existing established call
session at any time using functions
Hi,
Is it possible to define two pairs of network interfaces for rtpengine,
e.g. int/ext and up/down?
If so during a call established between interfaces 'int' and 'ext', is it
possible to send a re-INVITE from the calling party on interface 'ext', to
update the RTP destination to an IP on a
Hi,
I have been testing one Kamailio v5.1.6 instance with one rtpengine
instance, using sipp playing media files at 40 cps (-r 40) with up to 1600
concurrent calls. During the load tests if rtpengine is pkill'ed/restarted
a few times Kamailio would crash. It is quite repeatable and every time
. Since you can reproduce it, if I can test it
> here I can be sure the proper fix was done or the issue is somewhere else.
>
> Cheers,
> Daniel
> On 21.01.19 18:48, Yufei Tao wrote:
>
> Hi,
>
>
>
> I have been testing one Kamailio v5.1.6 instance with one rtpengine
org/errata/eid5294
Cheers,
Daniel
On 28.04.19 22:54, Yufei Tao wrote:
Hi,
I have a question regarding basic SIP flow for establishing a call session,
and wonder if anyone can help me clarify.
For normal call set up, INVITE-OK-ACK, from RFC3261, I believe the ACK’s
R-URI should be a c
Hi,
I have a question regarding basic SIP flow for establishing a call session, and
wonder if anyone can help me clarify.
For normal call set up, INVITE-OK-ACK, from RFC3261, I believe the ACK’s R-URI
should be a copy of the Contact header of the OK message.
However in RFC3665 sections 3.1
Hi Daniel and Victor,
Thank you very much for the very quick and helpful response, as always!
Best regards,
Yufei
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Hi,
I am trying to use KEMI lua that needs kemix module (for KSR.kx.<...>).
I wonder if there is any pre-built Debian package that include the kemix
module, something like kamailio-<...>-modules, that I can install? Or do I
need to build from source with kemix in included_modules?
On a related
Hi,
I got some problems with is_gflags:
1. Set the gflag:
# kamcmd cfgutils.set_gflag 1024
2. Check the gflags - fine:
# kamcmd cfgutils.get_gflags
0x400 (1024)
3. Check using is_gflag - problem:
# kamcmd cfgutils.is_gflag 1024
FALSE
Seems is_gflag always returns FALSE somehow. Calling function
33583a68a328aca9a3
>
> Cheers,
> Daniel
> On 19.03.20 15:01, Daniel-Constantin Mierla wrote:
>
> Hello,
>
> can you try:
>
> kamcmd cfgutils.is_gflag i:1024
>
> Cheers,
> Daniel
> On 19.03.20 12:12, Yufei Tao wrote:
>
> Hi,
>
> I got some problems wi
Hi,
The security tests were done to find theoretically possible flaws and help
make Kamailio "bullet proof". Well it's already a lot more robust than most
others. I think Daniel and Henning have made it very clear about the scope
of the bug.
For me if it is something that's been there for so
Hi Andrew,
Like Alex said, have a look at the commands RTPEngine received and the
responses it sent in the history of the call, and see if RTPEngine has ever
received any commands regarding this call, or, if the call has been deleted
from RTPEngine. You can raise the RTPEngine logging level to 6
Hi,
Like Olle said RTPEngine is perfectly capable of handling one leg in RTP
the other in SRTP. When looking up RTPEngine's documentation, may be worth
checking these out:
SRTP
SDES-unencrypted_srtp
SDES-unencrypted_srtcp
SDES-unauthenticated_srtp
e.g. try apply these flags to rtpengin_offer() on
Hi Duarte,
Not sure if I understand it correctly but are you trying to set up an HA
pair of dispatchers?
If so I am doing it differently in that, I use the Virtual/floating IP
(VIP) as socket. This way the master node (which has got the VIP) can send
OPTIONs pings out fine but the backup node
Hi Duarte,
I'm not sure if your situation is more complicated than mine. I have two
dispatcher nodes in the HA setup, with real IPs say IPa, IPb. They share a
floating/virtual IP, let's call it VIP.
Both nodes listen to the VIP (not its real IP), and use the VIP as the
dispatcher's socket. In
Hi,
RTP FO is possible, e.g. with keepalived+rtpengine+Redis as people
suggested.
Here's the guide for configuring it which I find very useful:
https://github.com/sipwise/rtpengine/wiki/Redis-keyspace-notifications
A possible way is to have a pair of rtpengine instances in master-slave
setup.
Hi Joel,
Thanks for the tip! Having the pings disabled and enable it on HA status
being master sounds like a good idea :)
Cheers,
Yufei
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Hi,
Here's an example kamailio.cfg that uses record_route so in-dialog requests
will go through the Kamailio proxy:
https://github.com/kamailio/kamailio/blob/master/etc/kamailio.cfg
There are many other example cfg files:
https://github.com/kamailio/kamailio/tree/master/misc/examples
Any server
Hi,
Here's an example kamailio.cfg that uses record_route so in-dialog requests
will go through the Kamailio proxy:
https://github.com/kamailio/kamailio/blob/master/etc/kamailio.cfg
There are many other example cfg files:
https://github.com/kamailio/kamailio/tree/master/misc/examples
Any server
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