Asterisk SIP Call audio delay

Asterisk Asterisk Support
1

System info:
Asterisk 1.4.28
Phones Polycom IP650 and Spectralink 8002

I’ve seen this posted in numerous other areas but no fix shown.

A good example of what we’re experiencing is described here:
mail-archive.com/asterisk@uc … 05938.html

However we’re doing SIP<->SIP not Zap<->SIP

When a caller calls in over a SIP trunk to our ITSP (Who’s using Metaswitch) the calls hit the IVR just fine, the caller can hear the IVR and can select options. However if they select an option to dial a ring group and we pickup there’s a 2-3 second delay in audio, the call shows connected, I can even see RTP flowing between the phone and the asterisk server but there’s not actual audible audio.

I’m at a complete loss on what’s happening here, it doesn’t make sense but it’s almost like asterisk is taking a second or two to sync up RTP between the ITSP and the phone (Because the IVR works fine).

If anyone has some insight on this please advise, I can share logs etc as needed but there’s nothing really to see because form a networking / packet flow standpoint it all looks right…

Thanks

2

Do you have a zaptel or dahdi installed in your system?

3

dahdi_dummy is installed (no physical hardware).

4

Yes, dahdi must give the clock. I think don’t exist an that was the issue. mmm.

How look the load of your COU when your audio fail?

5

[quote=“navaismo”]Yes, dahdi must give the clock. I think don’t exist an that was the issue. mmm.

How look the load of your COU when your audio fail?[/quote]

load averages are 0.00 across the board, it’s ~20 phones or so on a dual quad core Dell 2950 w/ 12GB RAM so we’ve got plenty of horsepower.

-SH

6

Did you get a resolution to this problem, I’m having the same issues?

7

If you need internal timing, you need not only a dahdi module (e.g. dahdi_dummy) but you also need enable internal timing in asterisk.conf, or as a command line option.