Grandstream GXP2000

gemini · Brekeke Member Joined: 11 Apr 2007 Posts: 18

1. Brekeke Product Name and version:

2. Java version:

3. OS type and the version:

4. UA (phone), gateway or other hardware/software involved:

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html :

6. Your problem:
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anyone else having toruble with brekeke software and grandstream gxp2000 phones?

trying to diagnose issues with call drops, I noticed they're not on the compatibility lists.

jelly · Brekeke Talented Joined: 20 Jun 2007 Posts: 62

i am using GXP2000 too, it works well with Brekeke PBX. what's your problem? let me see if i can solve your problem.

gemini · Brekeke Member Joined: 11 Apr 2007 Posts: 18

our ops group has a daily 10am conference call

the problems is the VoIP users drop calls every day,
almost exactly at the same time, time periods, every day
between 10-15 minutes into the call, then again
30-35 minutes into the next call, once rejoining the conferece call.
most often 90%+ of the time, the Voip user is muting the phone

I've looked at QoS, brekeke sip timings, multitech gateway timings, phone timings, I can't point my finger as to a cause of the drops, nort can I explain why on such a reguilar basis.

voipwell.com · Posted: Sun Aug 12, 2007 5:43 pm Post subject:

What do the sip logs say?

gemini · Brekeke Member Joined: 11 Apr 2007 Posts: 18

logs shows a normal call end or indicates a call success.

I got a call 'drop' on wireshark this morning, here is what I see

udp conversations, no flags set (Fragmentation OK)
between the phone, SIP server, and Gateway

Phone to SIP server
SIP server to Gateway
Gateway to SIP server
SIP server to Phone

near the 'drop'
I see 5 udp packes from the mutlitech gateway to the OnDo SIP Server same udp source port, same udp destination port, different data in packet, with no response from the SIP server sending to the phone, per established normal data transfer patterns, see above.

Then I see a normal BYE REQUEST from the sip server to the phone
and I see a normal BYE REQUEST from the sip server to the gateway. Call is done.

I see the phone reinitiate the connection to the gateway to as the user redials the number to rejoin the conferece call.

Something is causing a SIP BYE REQUEST

caller is on mute, may be happening during the transfer process from mute to no-mute. I cannot confirm or clarify. User is not monitoring call in such detail.

policy map on router interface shows no packet drops
Mutlitech gateways shows no packet errors
resource monitoring on server shows no heavy loads

gemini · Brekeke Member Joined: 11 Apr 2007 Posts: 18

I've been able to reproduce the problem.

* PBX is Nortel PBX with Digital phones, no VoIP

grandstream -> PBX extension, removes Nortel PBX from picture
call disconnect at ~9:00

grandstream -> softphone, removes gateway from picture
call disconnect at ~9:00

softphone -> PBX extension, removes SIP Server from picture
call ran for over 15 minutes.

issue point to GrandStream Phone

so, anyone with a grandstream out there could run a test for me?

on your GXP2000, call someone, put the grandstream phone on mute, with constent noise on the callee line, wait see if you disconnect at about the 10 minute mark.

Note call placed on mute not hold.

gemini · Brekeke Member Joined: 11 Apr 2007 Posts: 18

when on mute, some grandstream phones may not send RTP packets back to the server, this is confirmed via wireshark.

if the SIP server does not see any RTP packets, and the RTP session timer expires, the call is disconnected.

David Autumns · Brekeke Junior Member Joined: 02 Oct 2006 Posts: 8

Hi Gemini

I seem to remember there's a Keep Alive option within the GXP2000

It's not set by default

This may cure your problem

Regards

Dave