Painful audio problems with SIP connections


I hope someone can provide some insight to a show stopping issue we have been experiencing. We are running Elastix/FreePBX on a well powered dell server which is connected to the internet via. verizon FIOS 25up/25down. We have an office full of polycom 331 soundpoint phones on the LAN and 2 offsite phones of the same type. We are only using SIP service providers for in/outbound calling there are no analog lines are connected. We have tried three providers voipvoip, teliax (sip and iax2) and In every instance we have call issues about 5% of the time. Mostly this is garbled/clipped audio. Before you say network problems let me give a scenario that is hard to explain by network issues, latency, bandwidth:

We are on a call in the conference room with 3 parties in the room. 2 have dialed in directly from office phones – one of which is in the office and one of which is remote in a home office. The third party dialed via their pots line through the sip DID/provider. 5 minutes into the call The dialed in party becomes distorted/garbled. The 2 other parties are fine. The distortion lasts the duration of the call (40 more minutes). If this is a network issue why is the remote polycom not effected?

Similarly, there is never an issue dialing extension to etension inside or outside of the office. However there are frequent issues when dialing outside to a POTS line. It seems, again, that if network were an issue it would hit the remote polycom phones at least sometimes.

Again- this is with 3 separate sip providers that are all low latency and I rarely see dropped packets with wireshark analysis.

Calls are generally ulaw.

Thanks for ANY suggestions.


what are you using as the timing source?

My impression was that timing source wasn’t relevant when using pure sip connectivity i.e. no analog lines? I’m not even sure where to set that. Note… this problem exists in direct calls in and out as well as in the conference room.