RTP Relay failed

lakeview · Post by **lakeview** » Tue Dec 10, 2013 10:10 am

Did it happen during a phone call?
What SIP client and audio codec are you using?

voipwell.com · Post by **voipwell.com** » Tue Dec 10, 2013 12:48 pm

Also, it would be helpful to know how many minutes into the call the audio stopped.

farndt · Post by **farndt** » Thu Dec 12, 2013 2:47 am

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.

lakeview · Post by **lakeview** » Fri Dec 13, 2013 11:18 pm

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.

farndt · Post by **farndt** » Mon Dec 16, 2013 8:31 am

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

lakeview · Post by **lakeview** » Tue Dec 17, 2013 2:16 pm

Are you using own SIP client developed on SDK?

Does the problem happen with both Xlite and your SIP client ?

farndt · Post by **farndt** » Thu Jan 02, 2014 7:11 am

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.

james · Post by **james** » Sun Jan 05, 2014 2:00 am

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.

farndt · Post by **farndt** » Mon Jan 06, 2014 4:34 am

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

james · Post by **james** » Mon Jan 06, 2014 11:27 am

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?

farndt · Post by **farndt** » Thu Jan 09, 2014 2:58 pm

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)

james · Post by **james** » Sun Jan 12, 2014 1:42 am

You can expand the RTP port range at the [RTP exchanger] in the [Configuration]>[RTP] page.

Which OS are you using?