2 seconds no audio after pickup incoming call

Distro Discussion & Help
1

Hi Freepbx,

It is a very old problem, but still we are having this problem once in a while.
When an incoming call is answered, the caller can not hear the callee for a initial 2 a 3 second, after that all audio goes ok.

It does now always occur.

Also tested it with no firewall or nat.

When looking into documentation I found a setting in asterisk.conf: internal_timing = yes

I did not tested it yet, but could it be a solution?
I think it has something to do with rtp stream settings.

Does this problem occur to more people or is it a know problem? And is there a direction to look for in Asterisk or Freepbx to solve this or bypass this problem?

I hope someone can advice me or tell me some hints.

Many thanx in advance.

2

Important to be aware that call control (SIP) and call audio (RTP) are 2 different things. One is port 5060 UDP and the other is dynamic port 10000-20000 and usually UDP by default. You probably know that but important to point this out because therein is probably where your problem lies.

Also, if you have canreinvite=yes try change it back to canreinvite=no. Now called “directmedia” so look for either label.

3

Thanx Mustardman,

I already know that rtp goes over udp 10000~20000.
Also I know that reinvite could be a problem so I always set this to No.
Thanx for your tips but in that direction I already looked.
I seems that RTP stream is trying to connect but the call is already setup.
Tomorrow another day.
Thanx

4

You need to handle “early media” appropriately as yo how your VSP behaves.

5

Sorry Dicko, I don’t understand.
What do you mean with VSP?
The term early media is something I have seen in a Patton ISDN gateway.
Can I set it somewhere in asterisk or freepbx?

6

VSP = Voip Service Provider
"early media" is audio “rtp traffic” that is sent by the VSP before “answer” is given on that channel, apparently “. . for a initial 2 a 3 second” . . .

7

A quick workaround is to send the inbound call to an announcement that answers the call first with var/lib/asterisk/sounds/en/silence/1, then to your original destination.

1 Like
8

Wow Dicko that suddenly makes sense!
With this customer we have 2 different VSPs so that explains why the problem is not alway’s there.
I could be a VSP setting.
The workaround sound realy realy good and simple. I will try this, but it all makes sense to me now.
Thank you very much!!

9

I apologize in your case use /var/lib/asterisk/sounds/nl/silence/1. it is similar to the english version but more gutteral :wink: