28 Nov
2012
28 Nov
'12
3:05 p.m.
Hi,
From all of your support now I can call from
1.
IP Phone <--> IP Phone
2. Web Page <--> Web Page
3. IP Phone -> PSTN without any issue
But when I try to call from Web Page to PSTN then it tries to call
sip:00xxxxxx89078@mysipdomain.com and that time out.
Trying to figure out how to get this work ?
Can anybody guide me on this please.
Best Regards,
Roy.
28 Nov
28 Nov
5:02 p.m.
Unless your PSTN gateway supports the RTP/SAVPF media profile - I don't
know of any that do - this will not work.
Regards,
Peter
Hi,
From all of your support now I can call from
1. IP Phone <--> IP Phone
2. Web Page <--> Web Page
3. IP Phone -> PSTN without any issue
But when I try to call from Web Page to PSTN then it tries to call
sip:00xxxxxx89078@mysipdomain.com and that time out.
Trying to figure out how to get this work ?
Can anybody guide me on this please.
Best Regards,
Roy.
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Peter Dunkley
Technical Director
Crocodile RCS Ltd
6:48 p.m.
Dear Peter,
Thansk for your fast response. I highly appreciate it.
Is there any way that I can convert the RTP/SAVPF into general media
profile that PSTN GW support ? So that I can get that call working/
Best Regards,
Roy.
On Wed, Nov 28, 2012 at 8:32 PM, Peter Dunkley <
peter.dunkley(a)crocodile-rcs.com> wrote:
RTP/SAVPF
9:42 p.m.
On 11/28/2012 05:48 PM, Raj Roy Ghandhi wrote:
Dear Peter,
Thansk for your fast response. I highly appreciate it.
Is there any way that I can convert the RTP/SAVPF into general media
profile that PSTN GW support ? So that I can get that call working/
Best Regards,
Roy.
I think at the moment only a patched Asterisk might be able to do this:
http://code.google.com/p/sipml5/wiki/Asterisk
But I haven't got this to work yet because I get the feeling the patch
Doubango provides is incomplete.
Or maybe webrtc2sip by the same makers: http://code.google.com/p/webrtc2sip/
I've tried setting up a webrtc2sip server today but it crashes after a
few minutes, is horrible to set up (the web GUI is lightyears behind
Siremis) and for the moment I can't get it to work because I can't find
any proper documentation on how to set it up.
If anyone could build webrtc support into rtpproxy or any other media
proxy that can work together with Kamailio I'd be more than happy to
test it.
Regards,
Jeremy
9:58 p.m.
Jeremy, it is doesn't work at all. I've made a lot of changes to that
patched asterisk to make it working and no luck.
However, ast11 has fully supported webrtc, but I heard no voice during a
call.
Another issue is - sipml5 is sending a malformed Contact field, and
asterisk is trying to contact to invalid destination and finally closing a
call.
2012/11/28 Jeremy Jongepier <jeremy(a)autostatic.com>
On 11/28/2012 05:48 PM, Raj Roy Ghandhi wrote:
Dear Peter,
Thansk for your fast response. I highly appreciate it.
Is there any way that I can convert the RTP/SAVPF into general media
profile that PSTN GW support ? So that I can get that call working/
Best Regards,
Roy.
I think at the moment only a patched Asterisk might be able to do this:
http://code.google.com/p/**sipml5/wiki/Asterisk<http://code.google.com/p…
But I haven't got this to work yet because I get the feeling the patch
Doubango provides is incomplete.
Or maybe webrtc2sip by the same makers: http://code.google.com/p/**
webrtc2sip/ <http://code.google.com/p/webrtc2sip/>
I've tried setting up a webrtc2sip server today but it crashes after a few
minutes, is horrible to set up (the web GUI is lightyears behind Siremis)
and for the moment I can't get it to work because I can't find any proper
documentation on how to set it up.
If anyone could build webrtc support into rtpproxy or any other media
proxy that can work together with Kamailio I'd be more than happy to test
it.
Regards,
Jeremy
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10:14 p.m.
On 11/28/2012 08:58 PM, Konstantin M. wrote:
Jeremy, it is doesn't work at all. I've made a
lot of changes to that
patched asterisk to make it working and no luck.
However, ast11 has fully supported webrtc, but I heard no voice during a
call.
Another issue is - sipml5 is sending a malformed Contact field, and
asterisk is trying to contact to invalid destination and finally closing a
call.
Hello Konstantin,
Thanks for the heads up. Those sound like issues that could be resolved.
No audio or one way audio is almost always either a codec or a NAT issue
and the malformed Contact field is something I think could be worked
around too.
Regards,
Jeremy
11:50 p.m.
Am 28.11.2012 20:58, schrieb Konstantin M.:
Jeremy, it is doesn't work at all. I've made a
lot of changes to that
patched asterisk to make it working and no luck.
However, ast11 has fully supported webrtc, but I heard no voice during
a call.
Same here. I also tried the Doubango patch but it doesn't help.
regards
Klaus
29 Nov
29 Nov
12:57 a.m.
On Thu, Nov 29, 2012 at 2:42 AM, Jeremy Jongepier <jeremy(a)autostatic.com> wrote:
On 11/28/2012 05:48 PM, Raj Roy Ghandhi wrote:
I think it will be better for you to report the problem to
sipml5/doubango mailing list.
Many users reported that it works (including me).
Mamadou is a kind and helpfull guy.
Dear Peter,
Thansk for your fast response. I highly appreciate it.
Is there any way that I can convert the RTP/SAVPF into general media
profile that PSTN GW support ? So that I can get that call working/
Best Regards,
Roy.
I think at the moment only a patched Asterisk might be able to do this:
http://code.google.com/p/sipml5/wiki/Asterisk
But I haven't got this to work yet because I get the feeling the patch
Doubango provides is incomplete.
Or maybe webrtc2sip by the same makers: http://code.google.com/p/webrtc2sip/
I've tried setting up a webrtc2sip server today but it crashes after a few
minutes, is horrible to set up (the web GUI is lightyears behind Siremis)
and for the moment I can't get it to work because I can't find any proper
documentation on how to set it up.
If anyone could build webrtc support into rtpproxy or any other media proxy
that can work together with Kamailio I'd be more than happy to test it.
Regards,
Jeremy
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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Iwan Budi Kusnanto
4377
days inactive
4377
days old
7 comments
6 participants
participants (6)
-
Iwan Budi Kusnanto -
Jeremy Jongepier -
Klaus Darilion -
Konstantin M. -
Peter Dunkley -
Raj Roy Ghandhi