Wireshark mailing list archives
Re: Analyzing RTP Streams
From: "Boonie" <newsboonie () gmail com>
Date: Thu, 9 Sep 2010 21:41:09 +0200
Hi Dustin, If the playback proves that there is no problem at the point where you captured the packets but the internal user is complaining, move closer to that user. Maybe even use a SPAN port on the switch which has the user device connected. Dave ----- Original Message ----- From: Dustin Schuemann To: wireshark-users () wireshark org Sent: Thursday, September 09, 2010 2:06 AM Subject: [Wireshark-users] Analyzing RTP Streams I am trying to diagnose a VOIP issue. When I play the call it sounds fine. The call doesn't have any dropped packets. It does say payload changed to PT=0. The mean jitter was 1.29 ms, Max jitter was 12.64 ms. and the max delta was 22.93 ms. The internal caller hears the end users voice drop every other word. The outside caller doesn't have any issues. I believe this is a network issue but Im not able to pin point it. I guess I don't quite know how to read the RTP analysis screen. Any help you can provide would be grateful. ------------------------------------------------------------------------------ ___________________________________________________________________________ Sent via: Wireshark-users mailing list <wireshark-users () wireshark org> Archives: http://www.wireshark.org/lists/wireshark-users Unsubscribe: https://wireshark.org/mailman/options/wireshark-users mailto:wireshark-users-request () wireshark org?subject=unsubscribe
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Current thread:
- Analyzing RTP Streams Dustin Schuemann (Sep 08)
- Re: Analyzing RTP Streams Jaap Keuter (Sep 08)
- Re: Analyzing RTP Streams Boonie (Sep 09)