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VOIP Troubleshooting Issue

asked 2019-03-26 02:14:42 +0000

skoomabreath gravatar image

updated 2019-03-26 06:44:34 +0000

Jaap gravatar image

Having issues with VOIP call quality at one of my offices. I decided to take a simultaneous capture of outgoing/incoming RTP traffic between two of my offices. One of them has no issues with VOIP quality, and the other one does. I setup a port mirror at both sites before initiating the captures. The call sounded acceptable at both ends, but the Wireshark captures don't seem to match up.

Office #1 (no VOIP issues at this location):
RTP Packets Sent: 20,903
RTP Packets Received: 20,896
Timestamp Start: 16:27:36
Timestamp End: 16:34:34
Display Filters:
Outgoing Packets:
(ip.src == 192.168.7.30 && ip.dst == XXX.XXX.XXX.XXX) && (udp.port eq 2246 and udp.port eq 15286)

Incoming Packets:
(ip.src == XXX.XXX.XXX.XXX && ip.dst == 192.168.7.30) && (udp.port eq 2246 and udp.port eq 15286)

Office #2 (VOIP call quality issues at this location):
RTP Packets Sent: 9,533
RTP Packets Received: 9,517
Timestamp start: 16:27:31
Timestamp End: 16:33:53
Display Filters: 
Outgoing Packets:
(ip.src == 192.168.5.13 && ip.dst == XXX.XXX.XXX.XXX) && (udp.port eq 10066 and udp.port eq 10536)

Incoming Packets:
(ip.src == XXX.XXX.XXX.XXX && ip.dst == 192.168.5.13) && (udp.port eq 10066 and udp.port eq 10536)

Anything I'm missing here?

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Comments

It seems you have an intermediate gateway between the sites which terminates the sessions, and more importantly converting the media (hence the different RTP packet counts). Why is that?

Jaap gravatar imageJaap ( 2019-03-26 07:07:24 +0000 )edit

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answered 2025-11-20 13:10:49 +0000

gielo gravatar image

Just the following: The vital signs for voice are the following: Latency : Not exceeding 300ms round trip (max of 150ms one way) Packet loss: not exceeding 3%, preferably less that 1% (After 3% speech break up occurs) Jitter: max allowed is 20ms Bandwidth: G.711 (81.6 Kbps) average Bandwidth: G.729 (24.48 Kbps) average QoS for RTP = EF QoS for Signalling usually CS3 or EF

You do not mention what connectivity you have between the locations

From a Wireshark perspective I also had an experience where the call quality when listening to a call in Wireshark sounded worse then actual. This was caused when using the default setting of "Jitter" from the "Playback Timing" When I selected the "uninterrupted mode" from "playback timing" it matched and sounded the same as actual Hope this helps

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Asked: 2019-03-26 02:14:42 +0000

Seen: 565 times

Last updated: Nov 20