Hi, we are experiencing audio issues with 11 softphones recently installed at an international location and connecting via Internet.

Reported problems include noisy and lossy audio.

Initial testing of a single softphone worked well.

Client site and PBX site have ample Internet bandwidth. However Internet route traverses three other networks. Over the whole route there is inconsistant latency - several hops of 200+ ms where most hops are under 100ms.

I am trying to tune the connections using the IP settings for those softphones.

Does anyone have suggestions for the settings:

Avg frames/ IP packet
Avg in time frame % threshold
Avg in time frame timer
Minimum playback time
Speech encoding

I have a general understanding of what these do and a good background in networking but no direct experience with changing these settings.

Thanks for any help or advice received!

RTP across the Internet is not perfect. The 8602 is limited to 4 frames per packet in G.711 and 6 frames per packet in G.729, although I suppose lower values are tolerated. You can try adjusting the Minimum Playback buffer to a higher value, something like (average latency + (audio frames per packet * 10)). It won't be perfect but will get you close. Also, try G.729. Compression is your friend here. You can forget about the two Avg in Time Frame fields for now as they only serve to trigger alarms.

Thanks Jake!

I did realize after I posted about the avg in time frame fields only displaying a message.

I did switch to G.729B from G.729 and bumped audio frames up to 4. If there is no improvement today I will try your formula for minimum playback.

I don't know if the 8602's support G.729B.

After making those changes reports are that the softphones do not experience "slow audio". But still experience "scratchy audio".

I can't get a softphone user to call me so I don't know first hand what they are trying to describe.

I wonder what settings affect "scratchy" audio.

The thing to keep in mind is that the settings that you are tweaking are attempting to make the best of a bad situation. End to end delay should be below 80ms for "business quality" and below 40ms for "toll quality" or standard PCM quality. If it is fluctuating greatly you are down another performance benchmark as well by way of jitter which should be 20ms/40ms for the same quality.
-729B will save some BW with the VAD but BW doesnt seem to be your issue-delay is
-Upping your frames per packet will save on BW but again delay is your issue. Also upping the frames per packet makes loss more impactful if there is any.
The buffers can only compensate for so much. If you are expecting reliable good quality you will need to consider SLA from provider if they offer it (doubtful) or implement dedicated WAN connectivity to this site.

I've set audio packets per frame back to 3, sent the buffer to 120 ms and use BroadVoice for the codec. That seems to work pretty well.

The issue is that some days the route goes through Savvis and some days Sprintlink. With Sprintlink we see < 100ms every hop the whole way. With Savvis on some days the latency goes past 200ms and for some reason later hops on other networks also suffer as well.

We are monitoring the situation but yeah there is really not much we can do with that kind of network performance. We are planning to talk to the ISP's at the end points and see if they can think of a way to force the sprintlink route all the time.

Thanks for the advice and help!

No prob, good luck; true Internet telephony can be rough.

Broadvoice32 + Sprintlink = success.

The 3rd party vendor was able to convince their ISP to set a static route sending all their IP traffic destined for our network over Sprintlink exclusively. Sprintlink = low ping times.

That plus using the Broadvoice32 codec and tweaking up the received audio buffer a little bit cleared up our international VOIP issues.