Good to know about both -T and -W options. In many of my rtpproxy
installations I migrated to 2.0, because the default packaged rtpproxy
version from distros (mainly stable debian or centos -- which come with
some 1.2.1+patches), seems to be a problem with re-INVITEs changing the
RTP port.
Yes, Daniel is correct you can bump overall timeout with the -T option.
There is also another option -W now since 3 years ago (2.0+) to set timeout
specifically on early sessions, so that you can have bigger timeout during
call setup phase which might be relevant in this case. Hope it helps.
-Max
Hello,
knowing that rtpproxy v2.0 has -T is very useful. v1.2 is very old, so I
think its time to migrate to v2.0.
With this occasion I checked rtpengine -- it has a bunch of options
related to timeouts, so it's a good alternative to go for it as well.
Cheers,
Daniel
On 12.07.17 09:24, Marian
Hi,
works fine, after recompile RTPProxy with newer version
(2.2.alpha.20160822) and using option -T 150. There was no necessary to
modify the source code.
/usr/bin/rtpproxy -s udp:127.0.0.1 7722 -u rtpproxy rtpproxy -p
/var/run/rtpproxy/rtpproxy.pid -F -l XXX.XXX.XXX.XXX -d DBUG LOG_LOCAL0
-T
Hello,
ok -- it would be good to know if that solved the problem, so others can
find the answer in the archive when facing a similar situation.
Cheers,
Daniel
On 11.07.17 15:55, Marian Piater wrote:
>
> Hello,
>
> thank you for quick reply, I tried to find some options in the man
> pages, but
Hello,
thank you for quick reply, I tried to find some options in the man
pages, but I didn't read HOWITWORKS section.
I will try to compile RTPProxy from source.
Regards Marian
Dňa 11.7.2017 o 15:36 Daniel-Constantin Mierla napísal(a):
>
> Hello,
>
> doing 'man rtpproxy' on debian jessie
Hi Daniel,
we are using 1.2.1-2.1 version from Debian Jessie repository.
rtpproxy1.2.1-2.1
amd64Relay for Real-time Transport Protocol (RTP) media streams
Regards,
Marian
Dňa 11.7.2017 o 13:09 Daniel-Constantin Mierla napísal(a):
>
>
Hello,
I have a problem with RTPProxy.
Scenario:
SS7 (PSTN) ---> ASTERISK BOX --> Kamailio --> UAC
In case of incoming call from PSTN, when called party answer the call
after 60 seconds of ringing, then the call is without audio and in this
case I see in the log this message:
Creating session
