Intro to VoIP Telephony (Part 4)

Welcome back. Alright, last video. We're gonna leave you with a few tips for VoIP.

First one - If you have a cloud or hosted solution, and you can’t reach it with Q-SYS, try enabling STUN.

Then ask the provider if the domain is needed.

The username may not be unique and isn't always equal to authentication ID.

Certainly in Call Manager, it is not.

Alright, next one. In Cisco Call manager, there’s something called a media termination point.

Technically, it's no longer required but not in all cases.

There are a few edge cases where enabling it will get it working again.

In cisco call manager there's actually two different ways that you can set up a third party SIP device: basic and advanced.

With basic, you’re only enabling one extension.

So if you're using multiple softphones within a single Core,

you would have to use advanced, which allows up to 8 softphones.

If you need more than 8, you would actually use a SIP trunk, which allows for unlimited softphones.

When you're troubleshooting, take a look at the SIP traffic.

This will show us what device is sending the CANCEL or BYE messages.

And, once you know which device is stopping or preventing the call, we want to know why.

Usually, if the device is sending a BYE message mid-call for no reason it will provide an error code.

Feel free to do a search online for that code.

And if all else fails, and you can't find the code or you can't figure it out,

you should contact your administrator of that device or system to help you understand the why.

In most cases, it's the fault of some configuration on their side.

But at least we can get the customer pointed in the right direction whether it's on our side or theirs.

Enable softphone event logging if you want to do data gathering for troubleshooting.

Recall that if you enable logging within the softphone you're going to log all of the SIP traffic.

You're not getting any audio. Use this URL to find out: http:// - then your Core's IP address - /SIP.text.

It's gonna look like a plaintext website that shows all of your SIP logs.

Then if you want to get the network capture: go to http:///pcap_capture.html

Make sure the packet length is set to 1600.

Our engineer thought it would be useful for customers to be able to define that for themselves.

But in most cases that just caused problems because people would leave that blank,

in which case they're capturing packets of length zero, which means that you are basically capturing nothing.

You may need a successful call vs an unsuccessful call explained in detail.

Then you can also extract the logs as well.

You can also extract the system logs because there's SIP information in there too.

These are good tools that you can use for yourself, just to look at the SIP traffic and find out what's going on.

Alright, we made it to the end. Thanks for watching and we’ll see you next time.