| Author |
Message |
lakeview
Brekeke Master Guru
Joined: 15 Nov 2007
Posts: 319
Location: Florida
|
 Posted: Tue Dec 10, 2013 10:10 am Post subject: |
 |
|
Did it happen during a phone call?
What SIP client and audio codec are you using? |
|
| Back to top |
|
voipwell.com
Partner PBX
Joined: 20 Sep 2005
Posts: 528
Location: Tannersville, Pennsylvania
|
| Posted: Tue Dec 10, 2013 12:48 pm Post subject: |
|
|
| Also, it would be helpful to know how many minutes into the call the audio stopped. |
|
| Back to top |
|
farndt
Brekeke Member
Joined: 12 Jun 2013
Posts: 13
|
| Posted: Thu Dec 12, 2013 2:47 am Post subject: |
|
|
Hello,
It happened with 20 active sessions or more. The users have sometimes a 8h connection. but it happens after a view hours.
It Seems that there is a problem with the connection authentication.
Because there are a lot of 403 messages in the Error log and sometimes one IP gets blocked.
the used codec is g711a. |
|
| Back to top |
|
lakeview
Brekeke Master Guru
Joined: 15 Nov 2007
Posts: 319
Location: Florida
|
| Posted: Fri Dec 13, 2013 11:18 pm Post subject: |
|
|
Did it happen during a phone call?
What SIP client are you using?
> It Seems that there is a problem with the connection authentication.
SIP Auth and RTP are not associated. |
|
| Back to top |
|
farndt
Brekeke Member
Joined: 12 Jun 2013
Posts: 13
|
| Posted: Mon Dec 16, 2013 8:31 am Post subject: |
|
|
We use phoner, xlithe and a sip sdk.
The rtp worked till the call ends than no new connection was possible, so it seems.
The problem didn't reoccur untill now. But we are investigating it
try making a call when the sip register/auth fails  |
|
| Back to top |
|
lakeview
Brekeke Master Guru
Joined: 15 Nov 2007
Posts: 319
Location: Florida
|
| Posted: Tue Dec 17, 2013 2:16 pm Post subject: |
|
|
Are you using own SIP client developed on SDK?
Does the problem happen with both Xlite and your SIP client ? |
|
| Back to top |
|
farndt
Brekeke Member
Joined: 12 Jun 2013
Posts: 13
|
| Posted: Thu Jan 02, 2014 7:11 am Post subject: |
|
|
Yes both didn't work and the was reported today.
It seems that no rtp stream is coming from the tk to the sipserver.
but I don't have time to find out the system has to run.
I don't know when it happens and to log everything its to much data. |
|
| Back to top |
|
james
Brekeke Master Guru
Joined: 10 Dec 2007
Posts: 494
|
| Posted: Sun Jan 05, 2014 2:00 am Post subject: |
|
|
Make sure the network connection is stable enough.
Also you should use a non-evaluation edition of SIP Server if you want to the SIP server keeps running. |
|
| Back to top |
|
farndt
Brekeke Member
Joined: 12 Jun 2013
Posts: 13
|
| Posted: Mon Jan 06, 2014 4:34 am Post subject: |
|
|
Hello,
we purchased Brekeke SIP Advanced and it keeps running.
Only rtp seems to stop working. The connection is still there.
SIP still works fine |
|
| Back to top |
|
james
Brekeke Master Guru
Joined: 10 Dec 2007
Posts: 494
|
| Posted: Mon Jan 06, 2014 11:27 am Post subject: |
|
|
What kind of media stream was it? audio or video?
Which media codec was it?
Does the same problem happen even if a call is made between Xlite without your own softphone?
Which SIP SDK product are you using? |
|
| Back to top |
|
farndt
Brekeke Member
Joined: 12 Jun 2013
Posts: 13
|
| Posted: Thu Jan 09, 2014 2:58 pm Post subject: |
|
|
Thanks =)
I think I found the problem.
Brekeke seems to count the port-range up. 1, 2, 3, ...
But the media cards have a wide gap in their ranges
If the Ports reaches this gab no connections can be Established
So I have to separate the rtp port-range in two parts... is this even possible with brekeke?
(codec is standard g(711alaw) and we tried different sipphones) |
|
| Back to top |
|
james
Brekeke Master Guru
Joined: 10 Dec 2007
Posts: 494
|
| Posted: Sun Jan 12, 2014 1:42 am Post subject: |
|
|
You can expand the RTP port range at the [RTP exchanger] in the [Configuration]>[RTP] page.
Which OS are you using? |
|
| Back to top |
|
|
