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.
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
- IP Phone <--> IP Phone
- Web Page <--> Web Page
- 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. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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@crocodile-rcs.com> wrote:
RTP/SAVPF
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
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@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/Asteriskhttp://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/ 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@lists.sip-router.org http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**usershttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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
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
On Thu, Nov 29, 2012 at 2:42 AM, Jeremy Jongepier jeremy@autostatic.com wrote:
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.
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.
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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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Iwan Budi Kusnanto -
Jeremy Jongepier -
Klaus Darilion -
Konstantin M. -
Peter Dunkley -
Raj Roy Ghandhi