Delay in hearing the initial audio for 0.5-1 sec


#1

Hi

We are using Asterisk 15.4, sipML5, with bandwidth.com as our PSTN sip trunk provider.

For outbound calls we are not able to hear the initial 0.5-1 seconds audio (Like : Hi, this is, Welcome to, ) these things are cutting.

Can this be a Asterisk delay happening while connecting the two channels on bridge. Or it might be our SIP turnk provider.

Has anyone faced this problems before. Please let me known if anyone has solutions for this.


#2

Have you try with other trunk provider, also did you enable the rtp set debug on to chceck the media traffic while this happen ?


#3

As mentioned you really have to look at the over all RTP and the negotiation that is done on the WebRTC side. It may be taking time for ICE or DTLS to finish, which is needed in order to have media flow.


#4

Hi Jcolp,

As far as my understanding ICE gatherings/negotiations happen before the call is connected, correct me if I am wrong. But I am seeing choping in audio after answering the call.

Also, We have tested with other trunk provider (Twilio) as suggested by ambiorixg12. We didn’t see any choping there.


#5

That depends upon the SIP and SDP negotiation, so it may not happen until the call is answered.


#6

There are two ways to understand it. 1) SIP.conf asterisk IP sould not be 0.0.0.0 , make it fixed IP. 2) check RX and RX at system level for check message queue of server.


#7

0.0.0.0 is perfectly valid. It means listen on all interfaces.


#8

Hi all,

I have checked RTP packets with timings. I am seeing RTP packet flow immediately when the dialed number answers the call, with seconds level accuracy.

This bug is not consistent, we are observing in few calls. Can this be because the network is poor on one of the entities (Dialed side/ Asterisk / SIP truck providers / voip or PSTN). Since media will flow on UDP protocols, I am assuming UDP packets might dropped due to poor network at one of the entities. Or Is this a Asterisk system level issue, Has anyone faced this issue.


#9

You will need to analyse a failing case. Asterisk can be configured to log to millisecond precision.