Did it happen during a phone call?
What SIP client and audio codec are you using?
What SIP client and audio codec are you using?
RTP Relay failed
Also, it would be helpful to know how many minutes into the call the audio stopped.
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.
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.
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.
What SIP client are you using?
> It Seems that there is a problem with the connection authentication.
SIP Auth and RTP are not associated.
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
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
Are you using own SIP client developed on SDK?
Does the problem happen with both Xlite and your SIP client ?
Does the problem happen with both Xlite and your SIP client ?
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.
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.
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.
Also you should use a non-evaluation edition of SIP Server if you want to the SIP server keeps running.
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
we purchased Brekeke SIP Advanced and it keeps running.
Only rtp seems to stop working. The connection is still there.
SIP still works fine
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?
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?
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)
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)
You can expand the RTP port range at the [RTP exchanger] in the [Configuration]>[RTP] page.
Which OS are you using?
Which OS are you using?