Latency will not cause distortion, although continually changing latency may, if it forces jitter buffers to be reset.
Latency can exceed the capability of echo suppressors, and your latency is getting close to the level at which normal people cannot tolerate the round trip delay. (I’m assuming round trip. If this is one way, it is well beyond what an average person would find acceptable, and only really suitable for astronauts on the moon, where it is unavoidable.
It is your exceedingly high packet loss rate that is causing the distortion.
Actually If I ping the destination, there is almost no loss but yes latency varies between 5-15ms per second. Please correct me if I’m wrong but I believe varying latency is the cause of packet loss in Asterisk as there is sufficient BW, no packet loss, no CPU usage of server or any networking device.
Will implementing jitter buffer improve the quality without getting much delay?
Also I’m new to asterisk, if any one can guide me how to implement the jitter buffer? Do I have to just add the following command in extensions.conf or do I need to change other files to for that?
Varying latency will not cause packet loss of a type that would should up in the RTP statistics. SIP streams are handled as isolated packets. Even if there is a transition to circuit switching, and the jitter buffer overflows, the packet loss will occur on the outgoing side and won’t appear in the RTP statistics.
I would suggest running the ping with a large payload size and making sure you do a UDP ping, not an ICMP one.