Hi,
On Sat, Oct 8, 2011 at 2:09 AM, caio wrote:
> Hello,
>
> I have the following issue.
> A calls B, talk for a while, then B sends a BYE, but A never receives it.
> The BYE is transmited from the proxy listening interface:port to itself, but
> not to the asterisk port.
>
> U 2011/10/08 00
2011/10/11 davy van de moere :
> IMHO if I could convince Kamailio to always take the last part of the Route
> header into account and ignore the first one it would correctly work. What
> would be a good approach? Rewriting the route header looks abit harsch.
That does not make sense at all. The o
I'm trying to integrate to an integrics enswitch, almost everything works
like a charm, except on BYE packets as Kamailio in my setup forwards these
incorrectly.
Digging somewhat I believe the culprit sits in the Route header which comes
from the enswitch:
Route:
,
Kamailio takes into account th
2011/10/11 Juha Heinanen :
> i would prefer not to add more servers into my sip platform, because
> there is already sip proxy, registrar, presence server, xcap server,
> msrp server, and sems that provides several services. hopefully yours
> would replace some of those rather than add yet another
Iñaki Baz Castillo writes:
> No, it isn't. But OverSIP is not just a proxy providing WebSocket
> transport ;)
i would prefer not to add more servers into my sip platform, because
there is already sip proxy, registrar, presence server, xcap server,
msrp server, and sems that provides several servi
2011/10/11 Juha Heinanen :
> i read your slides and have question about the example: is OverSIP sip
> proxy still needed if Kamailio sip proxy/registrar would implement
> websocket transport?
No, it isn't. But OverSIP is not just a proxy providing WebSocket transport ;)
--
Iñaki Baz Castillo
inaki,
i read your slides and have question about the example: is OverSIP sip
proxy still needed if Kamailio sip proxy/registrar would implement
websocket transport?
-- juha
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sr
2011/10/11 David :
> If I understand what I saw correctly you have implemented a JS SIP stack
> which runs native in the browser over web-sockets.
Yes, it's a JavaScript code implementing a full SIP stack (parsing,
transactions, dialogs...) and uses a WebSocket connection for
sending/receiving SIP
Cool; I will wait until the code is available.
If I understand what I saw correctly you have implemented a JS SIP stack
which runs native in the browser over web-sockets.
On 10/11/11 8:11 AM, Iñaki Baz Castillo wrote:
2011/10/11 David:
This is really awesome.
Is any of that stuff on the v
2011/10/11 David :
> This is really awesome.
> Is any of that stuff on the video open to play with?
Not yet, but it will be.
> (I know it does not support audio but the SIP chat aspect is really great).
It's based on SIP MESSAGE. Implementing MSRP will be the next step :)
For audio/video we mus
This is really awesome.
Is any of that stuff on the video open to play with?
(I know it does not support audio but the SIP chat aspect is really great.
Thanks.
On 10/11/11 5:26 AM, Iñaki Baz Castillo wrote:
2011/10/11 Olle E. Johansson:
I think this is awsome. If I understand this right, t
2011/10/11 Olle E. Johansson :
> I think this is awsome. If I understand this right, this is based on your
> SIP over websocket draft, which basically adds a new transport to
> SIP. So for Kamailio, we would need a new transport.
>
> http://tools.ietf.org/html/draft-ibc-rtcweb-sip-websocket-00
Rig
El Tue, 11 Oct 2011 12:30:48 +0545
"Amar Tuladhar" escribió:
> Dear all,
>
> Hope you will help us with this issue.
Please, do not use the "answer" button to create a new topic. You break thread
views.
thanks,
jon
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SIP Express Router (SER) and
11 okt 2011 kl. 11:14 skrev Iñaki Baz Castillo:
> Hi folks,
>
> I would like to show a project my colleage José Luis Millán and me are
> working on. It's about SIP protocol running in a web browser. When
> RTCweb (media capable web browsers) becomes a reality both
> technologies together will al
Hi folks,
I would like to show a project my colleage José Luis Millán and me are
working on. It's about SIP protocol running in a web browser. When
RTCweb (media capable web browsers) becomes a reality both
technologies together will allow real SIP endpoints coded in
JavaScript running in a web br
On 09/28/2011 01:24 PM, Daniel-Constantin Mierla wrote:
Hello Marius,
On 9/28/11 12:06 PM, marius zbihlei wrote:
can you push it in a personal branch to review the impact on the
existing code. It is of interest, of course, ultimately there can be a
parameter (cfg or command line) to control i
Hey Alex,
On 10.10.2011 19:53, Alex Balashov wrote:
> Not only that, but the fix is not trivial. Contrary to how it may
> appear to a non-developer, this problem cannot be solved by just making
> a little patch.
>
> Stateless replies have that name for a reason; they lack state. They
> don't t
On 10/11/2011 02:45 AM, Amar Tuladhar wrote:
Hope you will help us with this issue.
In the INVITE message there is a field name CONTACT.
1.Is it possible that the ‘port’ in the CONTACT field is different
from the UDP port?
What's "the UDP port"? The source port from which the INVITE request
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