Troubleshooting Remote SIP

Troubleshooting Remote SIP Connections

Things to check if a remote phone or SIP Provider is not working. This can no audio, one way audio, poor audio quality, dropped calls, or other intermittent behavior.

When an issue is reported, we advise to start by ensuring the following settings are correct:

1. External IP or External Host are properly set in PBXSetup=>SIP
2. Extension is configured as a WAN phone (Under advanced in Edit Extension page).
3. ACL under Networking->Access Control->Access Control List make sure that the Remote IP address isn't being blocked.
4. The appropriate port forwards are configured in the Main Site Router Port Forwarding

• If you are troubleshooting a remote phone and you have a network available that you are certain works for registering remote phones to other systems, this network would be a good place to attempt to register the remote phone from. If it works, then the problem is most likely with the Remote Site Router as opposed to the router in front of the PBX. If it doesn't work then it's most likely an issue with the Main Site Router or and ACL issue.

• If you are troubleshooting a SIP Provider, try registering a remote phone and ensure that it works. This will help to discern that everything we expect to work with regards to the Main Site Router and PBX settings are correct.

• The PBX has a TraceRoute feature you can use to check for packetloss between the Main Site and the remote SIP public IP address. 0-5% dropped packets can bee acceptable, 5-10% can lead to poor audio quality, and anything greater than 10% may lead to dropped calls. By using TraceRoute or WinMTR you can demonstrate this loss to the end user and the ISP so they can work to resolve it. [Note] With sporadic issues you may need to test this multiple times, as there will be periods when it tests fine, other times it will show jitter.

• You can also use the following links to verify the bandwidth as well as packetloss/jitter at the Main and Remote sites. We advise allotting 100-200k up and down per call, any less and you could run into issues.

http://myspeed.visualware.com

http://myvoipspeed.visualware.com

A great resource for checking the SIP compatibility of a router, as well as instructions for changes that may need to be done on the router such as disabling ALG or SPI.

http://support.vocalocity.com/kb/router-compatibility-guide-and-recommended-equipment-list/

Phones Losing Registration every 2-5 minutes

Inconsistent NAT will cause a phone to lose registration at a specific interval, typically 2-5min. When this occurs, the phone will not be able to receive calls, but it can still make calls. If you are not able to replace the router with one that handles NAT more consistently, we have found that you can log into the phone and set UDP Keepalive Message to Yes. (If you are editing the Default Phone Template, the field will be called Keepalive, and needs to be set to 1 for enabled). If that does not work, you can also try to lower the Register Expire Time to stop the phone from losing registration. Both fields are found under SIP Account=>Account 1. We would advise to set the Register Expire Timer to a value below the interval you are seeing the phone lose communication. (eg. If the phone drops out every 2 minutes, set the value to 110 seconds, or tweak if that value does not fix the issue).