I have recently introduced SIP into our network and I'm having a hard time tracking down some intermittent phone quality issues. Before, I can track down my issues I need to understand why Wireshark is behaving the way it is.
when I run a capture on my SIP proxy box and open up Telephony->VOIP Calls and find a call that I'm interested in listening to. Sometimes when I click on the file and hit play stream the RTP player opens up but there is nothing in the box that shows source address port etc....so I cannot play the file why is that? Then I will find other calls that show codec is unsupported as there g729 calls. I found an article on WS showing the process to do this is that still valid as I'm running 2.1.1?
On my PBX there is a section where I can turn on voice stats and there are calls with a large number of packet loss. At the time there's no congestion on the WAN link or issues with the phone's LAN switchport so I'm trying to track down what's the root cause of my packet loss. I've got high loss with G711 and G729. I am new to SIP so if anyone has any pointers please let me know.
asked 06 Oct '17, 12:51
One, this is somewhat of a grab bag of questions, probably more suitable for wireshark-users mailing list, but here goes with some answers.
It may take a lot of drilling down into various system parts to get all blockages out.
answered 07 Oct '17, 01:21