Audio
Framesize
Sets the RTP packet size (msec Audio/RTP package).
Jitter buffer size
Specifies how many RTP packets can be cached to buffer or compensate for disruptions. (40 … 160 msec)
For more information, see Identities.Music on hold
If a connection or call is on hold, the "music on hold" is played.
You will find more information on this subject in the Music on hold topic.Codecs
Press this to display the codecs in the sequence in which they are used.
If you select a codec, you can replace it with a different codec, or none at all.
Best available quality: G.711, G.722
Best possible compression: G.726, G.729, iLBC
Note: Codecs that are assigned multiple times are automatically removed from the list and replaced with ---. You will find more information on this subject in the Identities topic.