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nhoague

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Hello.

We just installed Operator with 12 extensions. Seems every so often (very intermittently) outbound voice gets cut off.

This customer has a business class comcast, (20M/6M) internet so should be plenty of bandwidth.

I think I have all the services and ftp configured correctly. We are using QoS with 1M/1M.

Thoughts?

Here is a screenshot of the services.

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Vladimir Toncar (Kerio)

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You will need about 160 kb/s per call. Does the call end completely or is it choppy?

With most SIP providers, you do not need to map SIP and RTP to the local network. Sometimes yo might need to enable NAT support in Operator.

If you map SIP from any destination to Operator, you should use strong SIP passwords and use the built-in protection against password guessing. Do you do this to support mobile clients?
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vomsupport

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Check packet loss and latency..

We had a 10/50 with Comcast that continually dropped calls..

Only 4 extensions..

We moved it over to a 6 MB DSL and the dropped calls stopped...
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nhoague

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Can you tell me any other tools to check for loss and latency? I usually just use speedtest.net. We are using Nextiva as the SIP provider, and other sites have no problems. They have an online test tool and it always passes, even during times of dropped audio.

Thanks.
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PDigital

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We are also having the same exact issue. We have Comcast, Kerio 1.1.1 build 3918, updated Watchgaurd OS, and rebooted all networked devices. We use Bandwidth.com for SIP Trunk. I turned NAT on/off with no difference. When I call externally, the Kerio line can hear me, but I cannot hear them. Same issue vice versa. This issue is consistent. Also, when I call in from my cell phone, when the phone stops ringing it's supposed to go to voicemail, but I don't hear anything. But when I call from Google (internet phone) I do reach the voicemail. Weird right? We can call other extensions internally without any issues. The weird thing is, sometimes the internal line can successfully connect and establish a conversation on both ends. This however doesn't work all the time. When I restarted all the network devices again, I was able to make and receive calls successfully. After about 30 mins to an hour, this would fail, just like it is failing now. Any ideas?
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Vladimir Toncar (Kerio)

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It would help if you could send us a packet dump showing a call that is OK and then a call with the uni-directional audio. If you manage to record these two packet dumps, send them directly to me at vtoncar at kerio dot com.

Vladimir
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grazman

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the pcap file will show you the timing between packets. In general though you will want to look at whether your firewall has the ability to prioritize and shape traffic according to the ports (or sip protocol depending on how advanced the firewall is).

Some firewalls graph the internet use AND the quality of the internet connection (latency to their gateway). Example: with pfsense it will monitor the gateway. Problem is the gateway is the modem next to the firewall, so there's no real world latency measurement. So we usually use something like a google dns server (8.8.8.8 or 8.8.4.4) and have your firewall monitor and graph an alternate IP address as its gateway. This will help in a basic sense to draw a picture.

As for the bandwidth requirements, I find using the g711 codecs (most common), you need to allocate 83k of upload and download bandwidth per call.

Also, your watchgurad should have the sip application helper (sip proxy) turned off and the NAT ports configured like any firewall with asterisk behind it.
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PDigital

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No Message Body

[Updated on: Tue, 03 January 2012 22:25]

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grazman

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seems your reply was modded. I joined the forum today. I have a great deal of experience in troubleshooting sip related issues, and a very great deal of experience with firewall configuration and prioritization as related to sip.

simply put, I though putting the information on general terms here on the forum is appropriate. seems the moderator of the forum thought your response was not appropriate, to which I would agree.
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PDigital

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I actually modded my response. I thought this was a ticket that Kerio was working on. I apologize. When I realized this was a forum, I removed it.

It seems the problem was due to the fact the Level 3 Communications switched from using asynchronous platform to a synchronous platform. Which is weird since I signed my client for Bandwidth.com. I assume they use more than 1 carrier, which is fine, but I was given no notice. Bandwidth.com switched us to point directly to them, and since then we haven't had any major issues with hearing both ends.

I believe part of the issue was the Firewall rules may have not been setup right. I hired a Firewall/WatchGuard tech and the issue was resolved around the same time, so it's hard to say which exactly resolved the issue.

Just last week we had another drop call (could not hear caller (outgoing from office). I restarted the Kerio Operator server (not Kerio product) and started to work fine. Does you know if I can setup a rule to restart the server automatically maybe once a week? Or maybe avoid having to restart at all?

Thanks.
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grazman

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well, bandwidth.com also switched to the sonus switching platform, which has a lot of compatibility issues overall, especially as it relates to dtmf handling.

I actually moved my customers away from bandwidth.com entirely. They do/did a lot of things properly, but they downsized and considerably weakened their compatibility and compliance issues over the last 18 months. In doing so we've seen outages as well as certain call scenarios which never would function properly anymore due to their platform changes.

Typically outbound nat on the firewall needs to be set to use full cone nat, which is probably what your firewall tech enabled while disbaling the sip proxy mechanism on the firewall. At the same time if the udp port the audio was destined for was being used by another application, a port conflict could trigger a one-way audio issue. Usualy this is avoidable by using a separate/static ip address for your sip server though. resetting the states on your firewall or with the provider (assuming there are no other active calls in progress) would be another way to test it to see if it could be repeated.
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PDigital

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Thanks grazman.

Our Kerio Operator is using a static IP (internal static). Hopefully the new SIP company will not have these issues. We are moving forward with VoicePulse. They seem to have more Kerio specific support, but I guess time will tell.

Can I consult you in the future possibly?
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PDigital

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Finally moved to VoicePulse, but Kerio Operator is failing to register. We use a WatchGuard XTM 21. Working ok before porting from Bandwidth. I have entered in the user/pass/phone in Call Routing. I have the firewall policy configured. I just replaced Bandwidths with VoicePulses host name. I can see that Kerio is not able to register, and this is the same on VoicePulses side. They see the box attempting to register, but it does not.

<_at_>jfk-primary.voicepulse.com\' timed out, trying again (Attempt #103)


I am almost certain it's because I don't have something setup properly in the firewall. Any ideas?
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Vladimir Toncar (Kerio)

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It looks like the SIP responses from VoicePulse are being blocked. I would start with no special rules on the firewall. Our Voicepulse accounts work from behind NAT with no special firewall configuration.

Vladimir
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