I am trying to use jitsi for SIP connections. I am both trying with registered accounts on voiptalk.org (with and without their proxy nat.voiptalk.org:5065) and sipgate.co.uk. All call attempts to the test numbers or other phonenumbers fail with an error message "Call failed. The remote party has not replied. The call will be disconnected...".
Calling the same accounts with Linphone works fine - and Linphone uses STUN.
Has anybody got any luck with these providers and Jitsi ?
Any help is welcome,
Mark
Hi,
I've just tried by entering my SIP-ID@sipgate.co.uk and the sip password
and after that call the number 10000 and I successfully hear the
announcement.
Can you try recreating your account by simply entering those settings
without going to advanced options.
Cheers
damencho
···
On Fri, Jun 10, 2011 at 3:09 PM, Mark Dammer <clunymark@yahoo.co.uk> wrote:
I am trying to use jitsi for SIP connections. I am both trying with
registered accounts on voiptalk.org (with and without their proxy
nat.voiptalk.org:5065) and sipgate.co.uk. All call attempts to the test
numbers or other phonenumbers fail with an error message "Call failed. The
remote party has not replied. The call will be disconnected...".
Calling the same accounts with Linphone works fine - and Linphone uses STUN.
Has anybody got any luck with these providers and Jitsi ?
Any help is welcome,
Mark
Damencho,
I tried exactly that. Even with a completely clean jitsi setup (deleted .jitsi from my homedir).
No luck.
thanks for your help, Mark
···
On 10/06/11 13:38, Damian Minkov wrote:
Hi,
I've just tried by entering my SIP-ID@sipgate.co.uk and the sip password
and after that call the number 10000 and I successfully hear the
announcement.
Can you try recreating your account by simply entering those settings
without going to advanced options.Cheers
damenchoOn Fri, Jun 10, 2011 at 3:09 PM, Mark Dammer<clunymark@yahoo.co.uk> wrote:
I am trying to use jitsi for SIP connections. I am both trying with
registered accounts on voiptalk.org (with and without their proxy
nat.voiptalk.org:5065) and sipgate.co.uk. All call attempts to the test
numbers or other phonenumbers fail with an error message "Call failed. The
remote party has not replied. The call will be disconnected...".
Calling the same accounts with Linphone works fine - and Linphone uses STUN.
Has anybody got any luck with these providers and Jitsi ?
Any help is welcome,
Mark
Hey Mark,
Sending the logs [0] would probably help us be more specific but until
then, you could try disabling any codecs that you think you are not using.
A common reason for providers not to respond to calls is that they
simply don't get the INVITE-s. A common reason for that, on the other
hand, is that, unless you use TCP, the Jitsi INVITEs may often exceed
the MTU and go through IP fragmentation, which would be a problem for
some networks.
I guess that's the downside of supporting a large number of features.
Cheers,
Emil
На 10.06.11 14:56, Mark Dammer написа:
···
Damencho,
I tried exactly that. Even with a completely clean jitsi setup (deleted
.jitsi from my homedir).
No luck.
thanks for your help, MarkOn 10/06/11 13:38, Damian Minkov wrote:
Hi,
I've just tried by entering my SIP-ID@sipgate.co.uk and the sip password
and after that call the number 10000 and I successfully hear the
announcement.
Can you try recreating your account by simply entering those settings
without going to advanced options.Cheers
damenchoOn Fri, Jun 10, 2011 at 3:09 PM, Mark Dammer<clunymark@yahoo.co.uk> wrote:
I am trying to use jitsi for SIP connections. I am both trying with
registered accounts on voiptalk.org (with and without their proxy
nat.voiptalk.org:5065) and sipgate.co.uk. All call attempts to the test
numbers or other phonenumbers fail with an error message "Call failed. The
remote party has not replied. The call will be disconnected...".
Calling the same accounts with Linphone works fine - and Linphone uses STUN.
Has anybody got any luck with these providers and Jitsi ?
Any help is welcome,
Mark
--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
Jitsi
emcho@jitsi.org PHONE: +33.1.77.62.43.30
http://jitsi.org FAX: +33.1.77.62.47.31
Hi Emil,
you made this Sunday even sunnier: Both sipgate and voiptalk work now after I disabled a whole bunch of codecs. I am not an expert in voip codecs - could you recommend any codecs that always should be enabled or that are very useful in terms of quality bandwidth ? And on the other side what codecs are less important and more special ?
thanks for everybodys help here,
Mark
···
On 12/06/11 11:08, Emil Ivov wrote:
Hey Mark,
Sending the logs [0] would probably help us be more specific but until
then, you could try disabling any codecs that you think you are not using.A common reason for providers not to respond to calls is that they
simply don't get the INVITE-s. A common reason for that, on the other
hand, is that, unless you use TCP, the Jitsi INVITEs may often exceed
the MTU and go through IP fragmentation, which would be a problem for
some networks.I guess that's the downside of supporting a large number of features.
Cheers,
EmilНа 10.06.11 14:56, Mark Dammer написа:
Damencho,
I tried exactly that. Even with a completely clean jitsi setup (deleted
.jitsi from my homedir).
No luck.
thanks for your help, MarkOn 10/06/11 13:38, Damian Minkov wrote:
Hi,
I've just tried by entering my SIP-ID@sipgate.co.uk and the sip password
and after that call the number 10000 and I successfully hear the
announcement.
Can you try recreating your account by simply entering those settings
without going to advanced options.Cheers
damenchoOn Fri, Jun 10, 2011 at 3:09 PM, Mark Dammer<clunymark@yahoo.co.uk> wrote:
I am trying to use jitsi for SIP connections. I am both trying with
registered accounts on voiptalk.org (with and without their proxy
nat.voiptalk.org:5065) and sipgate.co.uk. All call attempts to the test
numbers or other phonenumbers fail with an error message "Call failed. The
remote party has not replied. The call will be disconnected...".
Calling the same accounts with Linphone works fine - and Linphone uses STUN.
Has anybody got any luck with these providers and Jitsi ?
Any help is welcome,
Mark
Hey Mark,
На 12.06.11 12:34, Mark Dammer написа:
Hi Emil,
you made this Sunday even sunnier: Both sipgate and voiptalk work now
after I disabled a whole bunch of codecs. I am not an expert in voip
codecs - could you recommend any codecs that always should be enabled or
that are very useful in terms of quality bandwidth ? And on the other
side what codecs are less important and more special ?
PCMU and PCMA are basically available everywhere. They also have good
quality even though they are narrow band but they are relatively costly.
G.722, Speex 16 and Speex 32 are wideband (often referred to as HD audio).
iLBC is quite low band.
Hope this helps,
Emil
···
thanks for everybodys help here,
MarkOn 12/06/11 11:08, Emil Ivov wrote:
Hey Mark,
Sending the logs [0] would probably help us be more specific but until
then, you could try disabling any codecs that you think you are not using.A common reason for providers not to respond to calls is that they
simply don't get the INVITE-s. A common reason for that, on the other
hand, is that, unless you use TCP, the Jitsi INVITEs may often exceed
the MTU and go through IP fragmentation, which would be a problem for
some networks.I guess that's the downside of supporting a large number of features.
Cheers,
EmilНа 10.06.11 14:56, Mark Dammer написа:
Damencho,
I tried exactly that. Even with a completely clean jitsi setup (deleted
.jitsi from my homedir).
No luck.
thanks for your help, MarkOn 10/06/11 13:38, Damian Minkov wrote:
Hi,
I've just tried by entering my SIP-ID@sipgate.co.uk and the sip password
and after that call the number 10000 and I successfully hear the
announcement.
Can you try recreating your account by simply entering those settings
without going to advanced options.Cheers
damenchoOn Fri, Jun 10, 2011 at 3:09 PM, Mark Dammer<clunymark@yahoo.co.uk> wrote:
I am trying to use jitsi for SIP connections. I am both trying with
registered accounts on voiptalk.org (with and without their proxy
nat.voiptalk.org:5065) and sipgate.co.uk. All call attempts to the test
numbers or other phonenumbers fail with an error message "Call failed. The
remote party has not replied. The call will be disconnected...".
Calling the same accounts with Linphone works fine - and Linphone uses STUN.
Has anybody got any luck with these providers and Jitsi ?
Any help is welcome,
Mark
--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
Jitsi
emcho@jitsi.org PHONE: +33.1.77.62.43.30
http://jitsi.org FAX: +33.1.77.62.47.31