Voice quility with Unity 4.03, CCM 3.3.3, and WAN links

[pbrown]

Hello All,

I have Unity UM 4.03 with 500 subscribers. It is integrated with CCM 3.3.3sr4a with 32 ports. There are 225 phones at the main site and about 400 phones devides among 13 remote sites. I have QoS working and voice quality is great between phones.

The problem - Voice quality is not at all good when listening to voice mail or greetings. I see that the phone is using G729 but jitter is all over the place when acciccing voice mail. BTW - jitter is low when speaking with another phone with G729 and WAN links. Unity was installed with G711 and I think that this is part of my problem. Transcoding is done on the CCM servers.

What should I do about this issue? I need a solution quickly.

Thanks,

Patrick

[Hin Lee]

Is the problem only with remotes phones accessing Unity?

Do you have TSP 7.0(3b)?

H. M.

[pbrown]

That is correct. The phones that are local to Unity (same LAN via G.711) work fine. Phones that are seperated from Unity by a WAN link sound poor. Sort of like bad cell phone service.

I may be confused about where the trancoding is taking place. When Unity was installed, I selected G.711 (I probably should have selected G.729). Phones at remote sites use G.729 when communicating with Unity.

Any ideas?

Thanks,

Patrick

[mrmccann]

Selecting G711 during install just means that the prompts and default greetings and voice names are G711 encoded. So this will force transcoding when calling across the WAN, which uses G729. I am not sure if the act of transcoding is going to cause the audio quality and jitter issues you are seeing. When you say 'jitter is all over the place', I thought of this DDTS: http://wwwin-metrics.cisco.com/protected-cgi-bin/ddtsdisp.cgi?id=CSCed52913

What packetization interval are you using on your CCM? As you can see from this DDTS, using 30ms and 60ms packetization with G729 will result in the phones showing high jitter, and in general using higher packetization interval, especially with G729, will result in audio quality degradation. If you are using >30ms and seeing problems, I would try using 20ms instead. This is set under Service Parameters for CallManager service - Preferred G729 Millisecond PacketSize.

You might also try checking out this doc for general audio quality troubleshooting tips:

http://www.cisco.com/en/US/products/sw/voicesw/ps2237/prod_troubleshooting_guide_chapter09186a00801d6d2a.html#1078488

[thomas.carbaugh]

Did you ever get this issue resolved? I am having the same issue with remote sites that are running g.729

[pbrown]

I did fix the problem. I inserted a protocol analyzer between the Unity server and the server switch (3750G) and again between the server switch and the core switch (4507R). It turned out that there was a bug in IOS on the 3750 when using auto QoS. The server switch was stripping the DSCP values in the egress of the 802.1q trunk. I ended up manually configuring QoS to get around the problem. I've since upgraded to 12.2.25 and I'm sure auto QoS works now but I have not tried it.

I also added a hardware transcoder for Unity and forced the Unity servers to G.711 only.

--Patrick