IP handsets drop mid conversation

[drewMk2]

Hi all,

has anyone had any issues with calls drop mid conversation?

my IP handset firmware version's are DBC 4422 R2E/R6C/R3H and 4425 DBC R2E/R6C/R3H.

"There is no place like 127.0.0.1

 

[dstrang1]

Bursty traffic on the LAN will cause that if you have no QoS or packet prioritization set up (or maybe what you have needs tweeking).

Good luck,

Dave Strang

 

[drewMk2]

Ok"...

hmmm QoS, but the strange thing is that it is only affecting a few handsets on each floor and extensions more then others. I have been investigating the DHCP server and the firewall.

we had an issue early in the install with the meda gateways and one way speach but this different now. when I do a call trace the there is nothing to trace where as before there was and now a network guy's are saying that all the faults are oin the MX.

does anyone have any ideas????

"There is no place like 127.0.0.1

 

[whosrdaddy]

what version of MX-ONE TS?

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[cv8devil]

I can answer that: MX-One 3.1 SP1

 

[drewMk2]

yes that right TSE 3.1 sp1

"There is no place like 127.0.0.1

 

[whosrdaddy]

these dropped calls, are those IP - IP?

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[cv8devil]

Calls are incoming external, outgoing external, internal IP.

in other words, all call types applicable to this site.

 

[whosrdaddy]

then you have a network problem. IP - IP calls are direct media, the MX-ONE is not involved during the conversation...

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[drewMk2]

arh ha,

this what I have susspected from the beging thanks for your help whosrdaddy.

cv8devil you may find this interesting!

"There is no place like 127.0.0.1

 

[whosrdaddy]

a simple ethereal trace on both sides (IP phone 1 and IP phone 2) should reveal what's going on...

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What You See Is What You Get
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[TBLuk]

A network problem is not 100% certain.

Do you have any data on the problem? i.e. is it always happening 'x' seconds into the call? Have you checked the QoS stats on the phone itself?
Have you upgraded to the latest phone firmware and compared results? (R6K)

 

[IPuser]

Isn't it the media that's diect media, while the call control is still via the LIM. If it's the calls actually dropping out - rather than just transmission being lost .
Signatl Trace on the call - to see where does the Clearing originate.

 

[TBLuk]

Yes, there's still a logical channel help up for call signalling.
Also since this problem is happening not just on VoIP to VoIP calls, it is worth finding commonality on the problem. i.e. time to disconnect etc

 

[whosrdaddy]

TBLuk and IPUser, I agree with both you, read my last post: an ethereal trace of both sides should reveal what is going on. When a call is ongoing, no traffic is going the the Server, only RTP and SRTP traffic between the phones. only if the user does something with the call then you have H245 signalling going to the server...

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