First do not use a mobile in the UK as the test. A normal landline in the UK to a mobile may show a high latency, but this is down to the mobile network.
You need to know what codec is being used, and what periodicity its working at. G729 only uses 8Kbs + UDP and IP headers. G729 in a connection with little or no packet loass, and consistent latency offers good voice quality. I have been on a site where the codec was changed from G711 to G729 one night and the users did not notice the difference.
Are you making test calls within the UK?
If Skype to Skype calls are OK then the latency to/from the Skype softswitch (or its media gateway) maybe OK. Another point is that a Skype to Skype call using SIP on the LAN may not go over the internet, it could just go up to the router and back.
If the problem is when calling UK landlines from the UK Skype the it suggests that the transcoding at the media gateway may be causing problems. Remeber calling a mobile will usualy mean trancoding to the UK landline network, followed by transcoding to the UK mobile network, hence my comment not to use UK mobiles as a test of latency.
A wireshark/tcpdump trace of the call can reveal many things, including the codec used, the protocal (SIP, MGCP or others) and by analysis the latency, and its variance. The trace may include the RTCP packets, and these show the delay, packet loss, ect.
Wireshark can decode the RTP stream and give more information. It can also output the RTP stream to an audio file that can be played on a PC.
You should also ask Skype for MOS/PESQ figures for their test calls to UK landlines. You will need to get to someone quite technical to give these figures, if they have them, and assuming they are willing to release them. For PESQ/MOS and G711 look for 4.0 or above. For G729 look for 3.7 to 3.8 or above. These figures are based on calls made within the UK, calls to/from other countries adds alot of other possible problems.
For VOIP calls the end to end transmission delay should be 150ms or less. VOIP calls still work OK above this, but 150ms is the standard figure given. Added delay is often due to the jitter buffer. The jitter buffer makes a massive difference to VOIP call quality.
I hate to say it but poor call quality on VOIP calls over the internet can be an absolute PITA to diagnose, and a major PITA to even try and get fixed. UK ISPs I know of do not honour the QoS marking on the IP packet. All the packets get treated the same, so if any part of the ISP's network is congested it will affect any VOIP calls passing through that congestion.
Rob
