Sometimes no sound on caller side ( NAT)

Asterisk Asterisk Support
1

Dear forum ,

I have a strange problem :

the Asterisk is behind my router inside the LAN which is nated…
I have configured my VOIP provider to receive sip calls. So far everything works fine,
i am registered and a external caller can establish a connection to the inside sip phone.

But sometimes the caller does not hear after attending the voice of the called one, just a contact noise is hearable if the called one speaks.

If the caller calls again, everything works fine again…

I would be happy if anyone can give me a tip what kind of problem that might be …

thanks a lot,

regards,

Filip

2

Which version of Asterisk are you using and which service provider? Also are you using the standard SIP port (5060/UDP) and which ports are you using for RTP traffic?

3

Hi Leemason,

its the version 1.6.0.1.
the RTP is reduced to 9078-9097 in the RTP.conf
The Service Provider is the german Dus.net. ( but I think with sipgate tis the same issue, if there are any doubts ill check that… ) The Sip Port to the SP external is of course 5060. But internal the Asterisk has to listen at 5061 b/c another serice uses 5060…

4

Do you have more than 10 concurrent, or recently completed, calls?

5

That seems a very small range for RTP. I believe, but could be wrong, that two RTP sessions are used for each call.

6

More precisely one RTP and one RTCP.

7

its in the rtp.conf:
rtpstart=9078
rtpend=9097

I didnt know that I can configure RTCP additional…

The system is used just for this test. Only one testcall at the same time, nothing else…

8

me again…

today i removed the RTP limitation. But the problem still apears…

anyone else for an idea?

thanks ,

Filip

9

Hello,

I know that this thread is a bit old, but was there ever a solution for that?
I have exactly the same issue with asterisk 1.8.10.1 on Ubuntu server 12.04 LTS with dus.net.

Meanwhile I’m a bit lost what else I could look at. I’ve also checked that the negotiated codecs are matching and they are the same for working and non-working connections.

Test wise I’ve set up a rtptimeout after 5 seconds and indeed the connections without audio are killed after 5 secs. But by what could rtp traffic occasionally be broken?

Any clue what can be done to find the root cause for this?

cheers
Tjareson