atcomsystems.ca/forum Forums TDM Business Telephone Systems Toshiba Telephones & Systems Temporarily cut dropped call still connected CIX670 : B--U CIX40A2 : GCTU2

No packet loss? You mentioned earlier that you had .49% packet loss. You might note that is 1 packet every 5 seconds on VoIP (G711/G729 send 50 packets per second). Additionally if your packet loss is .49% with ping at its defaults, you will most likely have higher packet loss once you start increasing the size of the pings (Use the -l option). Keep in mind that a G711 packet has a 160 byte payload.

Have you done a packet capture while the issue is occurring? Are you using anything to actively monitor and graph the latency on your network? Does it happen to all the remote users in the same office when it occurs?

mlaing,

Have you done a packet capture while the issue is occurring? yes
actively monitor and graph the latency on your network? not at this time.
Does it happen to all the remote users in the same office when it occurs? good question, it does happen too all users but rarly are they on the phone at the same time. on at least one occasion I know of it only happened to 1 of 3 people on the phone at the same time. the other 2 did not report it.

I am sorry I meant no loss while it was happening when we ran wireshark to catch it.
no resends, no fragmented packets or loss while it was happening.

we used the -l option I am pretty sure 256 byte payload

This is UDP. There are no resends. If the data is lost, it is gone. You wouldn't want to re-transmit a voice packet from 2 seconds ago, hence why they use UDP. Additionally if you are relying on wireshark to just tell you what the packet loss is, it won't. Have you had it analyse the RTP stream? Find the RTP stream, click on a packet from it and click Telephony->RTP->Stream Analysis.

Have you checked the latency across the network when the issue occurs? What you are describing sounds exactly like a network issue (Either high latency or packet loss).

mlaing,

Thank you for the help on this, we are now looking at the Telephony->RTP->Stream Analysis.

is layer 3 siwchtes needed for VOIP

I can't think of an advantage to having a layer 3 switch for voice. You can create VLANs with layer 2 switches. You could use layer 3 switches to route across VLANs, but usually the router handles that.

I don't know if you have the phones on a separate VLAN, but our company likes to separate the phones if possible on larger networks. The Toshiba phones support VLAN tagging.

I am pretty sure the problem is with Sonicwall router Model: NSA 2600 Firmware Version: SonicOS Enhanced 6.1.2.3-20n