Re : Call Quality

[mitelphantom]

I have a big problem whereby, calls are dropping and call quality is poor over sip trunks, i have enabled g729 compression at the sip providers end and have enabled 16 compression licenses and isntalled a dsp module but still my customer has call quality issues. In network element i have my sip provider in zone 2 which means inter-zone compression should work because my phones are in zone id 1, But they are dnic sets and not sure if this makes any difference. Any previous experience with this sort of issue is highly appreciated. Also does is there any difference if the users happen to use headset, speakerphone, or handset if they dial out using sip trunking ?

 

[IrwinMFletcher]

You'll need to be a bit more specific about what you mean by 'poor quality'. Does it just not 'sound right', or are there audible gaps in the speech? Echo? Distorted words?

There is no difference in how the call gets set up in terms of heaset, handsfree, etc, but there is a significant difference in the call quality since different physical speakers/microphones are involved.

One other point, enabling G729 will only help in a low bandwidth situation. G729 is a compression algorithm so it actually reduces the overall call quality. Where it helps is if the bandwidth is low enough that G711 calls would drop packets.

 

[MITELPETE]

I have had no experience with DNIC's I'm afraid.

I have had call quality issues on (Global Crossing) Sips if the Compression licences haven't been added AND the system rebooted to activate them.
I tend to put the phones in Zone 2 and leave analogues in the same Zone as the SIPS (essential) so that no compression is carried out and the call stays at G711. If the SIPS are in Zone 1 then they are in the same Zone as the Mitel which may be the winning combination. Zone 1 - Mitel, SIPS, Alogs. Zone 2 - IP Phones

If your SIP provider is Thus check the size of packets because they might operate at 10 ms packets rather than 20 ms packets(that Mitel operates with)

 

[IrwinMFletcher]

MitelPete, you must have had bandwidth issue then I take it?

A SIP provider that uses 10ms will not work when talking to the 3300 (analog endpoints, TDM trunks, conferencing, MOH, etc), it may however work with the IP sets (not sure) but even then the sets will transmit 20ms so the SIP provider will have to handle that. The ptime negotiated in SIP is only a recommendation, the endpoints are free to transmit something different.