Asterisk sip show channelstats

hello everyone, I got this response by run this command in asterisk CLI “sip show channelstats” with my data as well, can anyone help me by telling me which is Jitter & which is RTT from below data.

Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter 0bb9262d6a1 00:01:12 0000003558 0000000000 ( 0.00%) 0.0000 0000001177 0000000000 ( 0.00%) 0.0200
1 active SIP channels

I don’t see your problem. The number of field labels matches the number of fields, so the receive jitter is 0 and the send jitter is 0.02. There is no round trip time recorded.

Thanks for your fast reply, if there is no round trip time recorded, how can I get average latency of each channel? is there any to get the average latency realtime asterisk channel

You can’t because it is an end to end thing and the best you could do would be hop by hop.

Also RTP time stamps are not well enough controlled to allow them to used to split RTT into latency in each direction, and there is not RTP echo function, so you cannot measure latency in the actual RTP stream by looking at only one end. Other protocols, like RTCP and SIP will be prioritised differently on any network specifically designed tor controlled latency. SIP tends to need high reliability, whereas RTP can accept more loss, to achieve a lower latency.

Well, we have done already by splitting RTT into latency & its working very fine & we are getting the data after channel hangup, but now I want to build it in real-time, I hope there is some way & I will found.