, 2015 at 7:26 PM, Rahul MathuR
wrote:
> Hello Vasiliy,
>
> Thanks for replying.
>
> Not sure why Linphone-Android won't receive INVITE, since it is responding
> well to the keep-alive OPTIONS messages from proxy.
> Both, proxy and SIP server are sending packets to UAb on
Hello Vasiliy,
Thanks for replying.
Not sure why Linphone-Android won't receive INVITE, since it is responding
well to the keep-alive OPTIONS messages from proxy.
Both, proxy and SIP server are sending packets to UAb on which UAb is
apparently responding to only proxy.
Is this a genuine flaw in
Hello Daniel,
Thanks for replying back !
And please accept my apologies for responding late. I tried modifying the
configuration file to do a far-end NAT traversal but this case (wifi to 3G)
is still not working.
Below is the trace of what is happening on my systems -
Registration of UAC Behind
ifferences are the way that we detect a
> video call, how we route to our backend servers, and that we send video
> calls directly to a registered peer and not the the backend Asterisk
> servers.
>
> On Thu, Feb 12, 2015 at 12:34 PM, Rahul MathuR
> wrote:
>
>> Gentle Remin
Hello Manuel,
To support the hypothesis of crypt libs screwing the logic, you can try a
'secure call' without using webrtc.
If them are to be blamed; your 'secure call' won't be successful.
Aside this, you can get a better idea of what has dwell-ed behind the
curtains by looking at syslogs.
Hop
Gentle Reminder !
Thanks
Warm Regds,
Rahul
On Thu, Feb 12, 2015 at 12:13 AM, Rahul MathuR
wrote:
> Thanks guys !
>
> I did further investigation of the Chrome logs and found that... (this is
> really interesting), even though I disabled Video; still JSsip was sending
> video
;
> I corrected the other issues I was having and that one seemed to resolve
> itself.
>
> Hope that helps,
> Marc
>
> On Tue, Feb 10, 2015 at 12:01 PM, Rahul MathuR
> wrote:
>
>> Hello gents,
>>
>> I was trying my hands on getting a successful RTCweb ca
Hello,
I was wondering whether Kamailio (as proxy) can generate a PRACK on its own
( since one of the custom written dialer is not sending PRACK) ?
Is there any way I can achieve this ?
--
Warm Regds.
MathuRahul
___
SIP Express Router (SER) and Kamai
g sample config using the following architecture:
>
> https://github.com/spicyramen/llamato/tree/LlamatoReg
>
> signalling: sipml5 -- ws/wss --> Ec2 Kamailio --sip udp--> FS --sip
> udp--> *
> media: sipml5
> ---
ue, Jan 27, 2015 at 2:14 AM, Rahul MathuR
wrote:
> Hi Richard,
>
> Thanks for spending some cycles on it.
>
> It is OpenSSL 1.0.1e-fips 11 Feb 2013
>
> On Tue, Jan 27, 2015 at 2:04 AM, Richard Fuchs wrote:
>
>> On 26/01/15 02:21 PM, Rahul MathuR wrote:
>>
>
Hi Richard,
Thanks for spending some cycles on it.
It is OpenSSL 1.0.1e-fips 11 Feb 2013
On Tue, Jan 27, 2015 at 2:04 AM, Richard Fuchs wrote:
> On 26/01/15 02:21 PM, Rahul MathuR wrote:
>
>> Hello,
>>
>> I am totally struck at a point while implementing Kamaili
route for rr module ).
>
> My 2 cents.
>
> Le 26/01/2015 11:21, Rahul MathuR a écrit :
>
> Hello,
>
> I am totally struck at a point while implementing Kamailio as proxy for
> WebRTC enabled UAC (Jssip). I am using Google's TURN server
> (rfc5766-turn-server for ICE
Hello,
I am totally struck at a point while implementing Kamailio as proxy for
WebRTC enabled UAC (Jssip). I am using Google's TURN server
(rfc5766-turn-server for ICE/STUN). I am able to get to the point where the
SIP server sends 183 session in progress to kamailio but after that I can
only see
Sat, Nov 29, 2014 at 2:16 PM, Daniel-Constantin Mierla wrote:
> Hello,
>
> upgrade to use the latest stable version from 4.2 branch, respectively
> 4.2.1 at this time -- you are running a pre-release version, which was not
> supposed to be ready for production anyhow, and there was
Hello,
I have recently moved some traffic to kamailio-4.2 (installed on a seperate
box) and it crashes quite often. Below is the backtrace of the core files -
(gdb) bt
#0 0x0033d8c32635 in raise () from /lib64/libc.so.6
#1 0x0033d8c33e15 in abort () from /lib64/libc.so.6
#2 0x
Hello,
I'm new to WebRTC although I've been using kamailio as sip proxy server for
few months now. What I really do not know and trying to understand is -
a) Can kamailio be used as sip-proxy while using WebRTC based UA calling to
plain UAC/WebRTC based UAC ?
b) What to use for media proxying (t
Hello,
We have observed a strange behavior in corex module that it gets loaded at
every sip packet which arrives to kamailio.
We put a static variable and saw that it gets re-initialized to 0 everytime
any sip packet comes to it.
Could you please tell me how to stop it and load it just once.
Th
Hi,
Did you get some free cycles to look at it ?
On Wed, Sep 17, 2014 at 12:12 AM, Rahul MathuR
wrote:
> Thanks for replying !
>
> But how to check whether a particular message received by Kamailio was
> sent by UAC or SIP Server ?
> Also, on the same lines - how to know wheth
dp125704
>
> You can also use any other kamailio language bind of you choice as well,
> e.g. Python, LUA, JAVA and so on.
>
> I would recommend the second option, as it has less processing overhead
> for kamailio.
>
> Thank you.
>
>
>
> On Tue, Sep 16, 2014 at 6:09
Hello,
I was going through the new features and stumbled upon this new one -
developed by Mohd. Shahzad Shafi.
As already mentioned on the wiki about this module, I intend to use it for
my custom security layer between UACs and SIP Proxy (Kamailio) but the
issue is - the custom security layer (enc
Thanks for the help !!
It is now resolved.. my special thanks to Davy !!
On Thu, Aug 14, 2014 at 7:12 AM, Rahul MathuR
wrote:
>
>
> Hello Davy,
>
> My constraint is that I cannot use TCP for this solution.
> I am attaching my kamailio.cfg file, please please help me to re
together with the logic which went before
>> it…
>>
>> A good old tcpdump will most likely enlighten us.
>>
>> Op 12-aug.-2014, om 14:39 heeft Rahul MathuR
>> het volgende geschreven:
>>
>> Hello Davy,
>>
>> Thanks for writing back..
>&g
ten us.
>
> Op 12-aug.-2014, om 14:39 heeft Rahul MathuR
> het volgende geschreven:
>
> Hello Davy,
>
> Thanks for writing back..
>
> Tonight I'll take the tcpdump on Kamailio box and share the file.
>
> Please note that Kamailio and Freeswitch are both on pub
r 30 seconds disconnects, as for FS the
> call has failed.
>
> Do you have a trace of the packets?
>
> grtz,
> Davy Van De Moere
>
>
> 2014-08-12 13:37 GMT+02:00 Rahul MathuR :
>
>> Hello,
>>
>> I have an iPhone/Android/Windows 8 based UAC, proxy serve
Hello,
I have an iPhone/Android/Windows 8 based UAC, proxy server Kamailio and Sip
server FreeSwitch.
Whenever I call directly from UAC to Sip server, the call gets established
for as long as I want, however when I use the proxy in between, it gets
disconnected within 30 seconds. It seems that FS
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