Advice for an audio quality issue ...


Jul 17, 2009
Hello all.

I invite you all, please, to give me your general opinion of the bother I'm having.

I took a bold step about 3 months ago and began recommending VOIP as a replacement phone service for 3 offices. I had been getting deeper and deeper into Asterisk/Trixbox/Elastix over the last year or so, using a small non-production setup of my own to hone it.

There are three offices suffering this altogether but I shall use one as the example of what the users have been, and still, experiencing;
This particular office has a 10mb Fibre internet connection (thats 10mb both ways) terminated by a Draytek 2930 Firewall, a dedicated Core2Duo workstation sitting on the LAN with 2gb RAM running Elastix Each desk houses an SPA942 LinkSYS handset connected to dedicated PoE switches (around 40 handsets in all).
I have configured an IAX2 trunk to VoIPtalk and set up IVR's, Queues, Ring Groups, Pickup Groups, Day/Night, Time Groups, and even found time to update openfire to the current release and get Spark on machines.
I have perfected the Endpoint Configuration Template within /var/www/html/modules/endpoint*/libs/vendors to includes full descriptions of both sections (MAC and Template CFG generation) and to include the GMT time zone, NTP, Busy Lamp notification (taking advantage of the current SPA942 6.1.5a firmware) and have duplicated all of this at another site and linked them both together via VPN using IAX2 Trunks. Frankly everything on the network working very well indeed.
Internal calls rock, not a glitch. Wide area calls to the other office are superb.

Then... they make/receive external calls and it is aweful;

- Speech delays are noticeable of around half a second (occasionally more)
- A caller and the recipient cannot speak at the same time, one or other of them gets muted out or their volume gain reduces considerably
- During a call, the conversation will "freeze" for up to 8 seconds (the longest recorded) and then re-establish audio and allow it to continue
- Various quantities of snapping, crackling and popping can be heard in the background during conversations.

So ... they have a dedicated 10mb Fibre internet connection, a seperate gateway for the VPN to the other office so as not to interrupt the 10mb Fibre, instant calling when a number is dialled (no delays when contacting the server) and external phone calls roll in without issue, 100% of the time.

There isn't any QoS on the 10mb Fibre as nothing else uses it for the time being; similarly no QoS on the switches as they only have phones on them.

Could the first item be codec related?
Could the second item be, again, codec related or is there a "Duplex" setting in Asterisk, or "Auto Volume Level Sensor" in Asterisk which is causing this?
Could the third be the UDP packets connecting to getting fragmented and having to re-sync in some way?
Could the fourth be, again, codec related but only for external calls (no audio issues as yet have been reported for internal calls)

I am desperate for all of your opinions as I have now exhausted my search for answers and VOIPtalk appear to be stumped.

Many thanks in advance.

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