sundance-communications.com/forum Forums TDM Business Telephone Systems Toshiba Telephones & Systems Intermittent Call Quality Issues on Inbound Calls (SIP)

CIX 1200

We are currently using SIP trunks for our inbound traffic and an ISDN PRI for outbound. Our outbound calls are great and there are zero issues. However, we are constantly dealing with call quality issues on our inbound SIP trunks. The issues are intermittent making it very difficult to troubleshoot. There doesn't seem to be any pattern other than that the longer the call goes on for, the greater the chances that the quality will degrade over time. I've ran a wireshark and did find some packet loss and jitter, but I'm not sure why that is or how I would resolve that. I'm thinking that I should probably play around with the Voice Packet tables and/or try changing the audio codec, but I just don't want to introduce any issues to outbound calls since they are working fine. I don't believe it to be a bandwith issue either since the outbound calls are fine. Any suggestions are appreciated!

I'd also like to add that the problem is always that the caller on the other end is unable to hear us, but everything sounds fine on our end. To the outside caller, the voice sounds choppy and they only hear about every fifth word. Sounds like a classic case of jitter to me...

Also, when I listen to the calls in Oaisys Tracer (call recording), the calls sound 100% clear. Very strange!

Sounds like a Carrier issue.

you should run wire shark and have toshiba engineer take a look

You did not mentioned what type of Internet you are using for your SIP trunks. You said you don't think it is a bandwith issue becasue your outbound is ok, but your outbound is PRI correct? Do you have a bundled data/voice over T1?

since callers have trouble hearing the local caller, it would sound like an upload issue, either bandwidth or QOS. Your Oaisys recording is clear because it is listening locally, not through the internet like caller on the other side of the SIP trunk.

Not much you can do with packet loss. Lost packets means lost audio. You culd try to adjust jitter buffer, but I would not put too much faith in this helping much. These pretty much point to your ISP. I would expect this more from DSL then T1.

We are on a 10MB Fiber Ethernet, Switched network. And yes, we are using a converged network for data/voice. I'm thinking we may need to implement QoS as data/voice packets may be fighting for priority.

The easy way to tell is the difference in voice quality depending on the time. If it works great after hours, when there's no data running (and competing), then QOS is going to be the answer.

Quite frankly, on ANYTHING that's not dedicated entirely to VOIP, I shoot for QOS. It can never hurt, and will certainly improve poor situations.