Good afteroon all,
First of all: thanks for everybody who is willing to think this through with me:
I’m having some issues regarding call quality between some calls. Let me try to explain the situation first
We have a Asterisk 11.16 server based on the Xivo distribution. There are 2 servers running in cluster (Active Passive), both virtual with the following config:
8 GB ram About 50Gb of diskspace which is used for about 15%
(Let’s call this Asterisk cluster 001 for clarity)
The Asterisk server has a trunk to a cisco call manager which is on the same site/LAN, and 4 trunks to other Asterisk servers (same distribution but lower specs, name Asterisk cluster 002 and 003). These are all sites in our WAN but they are geographically divided and connected via MPLS links. Each affiliate has a specific number range XXXYYY where XXX stands for the affiliate and YYY is the extension of the users.
(Average bandwidth = 4Mpbs which has to be shared by applications. QoS allows that VoIP is prioritized)
Now, the actual problem:
I’ve set my main codecs to G711 a-law, G7 222 (for cisco call manager) and GSM as last. The GSM is set as primary for those trunks which don’t have 4 Mbps of bandwidth available.
In most cases, trunk calling results in bad quality of conversations (a-law is chosen as codec) but or it is jitterish, or one party does not hear the other party (complete silence) It could be that the second time they call, everything is ok.