Intro to VoIP Telephony (Part 3)

Welcome back! I promised you all that we’d talk about DTMF, and gosh darnit we’re gonna do it. We're gonna do it guys.

DTMF is used for sending numbers and issuing commands when dialing.

For example, 697 Hz and 1209 Hz are the two tones that make up the number one.

There are two types of tones, there's in-band and out-of-band.

So when we're talking about in-band we're talking about RTP streams,

and out-of-band means that the digits are sent in SIP messages instead of RTP.

There are two versions that we support within the Q-SYS softphone: RFC 2833 and DTMF info.

RFC 2833 is the SIP standard sending DTMF, while DTMF INFO is old and not really used anymore.

DTMF in-band shares the same RTP streams with the DTMF tones as voice. It's often heard by all participants.

So if a customer says, “hey I don't really want to hear all the button presses and all that stuff,”

well you can't really use DTMF in-band.

Now there is no formal in-band DTMF tones built into Q-SYS but you can generate your own DTMF tones

you can embed into your own audio traffic if you need to.

Just build the tones into a sine generator and expose the proper pins to trigger them.

Then you'd mix the DTMF into your microphones, and then send them to your Q-SYS VoIP out component.

And then you would use your VoIP IN component block to receive the far end audio

and send those to the loudspeakers.

The VOIP status in the control block is relatively simple,

which acts as a tone controller for all these different tones.

In this example we've got the DTMF playback enabled and we've got a ringtone enabled.

We also enabled entry and exit tones. These are the tones that you do that with.

If you notice, this is going to be a single tone because it's a single file.

This is because if you put an actual DTMF tone in there there's a chance that you'd get a double detection.

If you're actually hitting the buttons and it's playing back on the loudspeaker

you could be detecting that as a DTMF in your own system as well.

Not really what you want, right?

It makes more sense to just use a different tone that you know is not going to match a DTMF tone.

This is the status and control block, where you’d push all the DTMF buttons while you're in the call.

Beyond the basics, we do have the do not disturb, connect, disconnect, off hook, ringing, and turn on auto answer.

You’ll also see the status of the softphone, which is communicated by the SIP.

We also have this option for “continual DTMF” and “simulate incoming calls” which are for troubleshooting.

This screen is the softphone Configuration screen from Core Manager.

At the top, you can specify which LAN you want the Core to use for VoIP.

Generally speaking, most installs will use LAN A for Q-LAN audio

and use a LAN B to connect to your enterprise connection and VoIP.

However, this is not a hard-and-fast rule. You can use LAN A or the AUX port for VoIP. It’s really up to you.

5060 is the standard port for SIP so many times you don't have to change that.

Different solutions may require you to use a different port.

Like Ring Central, for example, uses 5090.

Below that, you can enable DTMF info, which is set to “no” by default,

but you would need to enable it if you are using an Avaya system.

Next is RTP Type which is usually 101.

When you set your Core to Stun, you’ll receive immediate praise from James Tiberious Kirk himself,

and also that sentence is utter nonsense.

Enable Stun to traverse NAT if you are behind a firewall.

So if you have a hard time reaching outside the network or the enterprise firewall, you can always turn that on.

Generally you would “Enable SRTP” traffic if it's a government client or something like that.

Here’s where you're going to enable your preferred audio codec

and the order that you want them to default to.

Use the up and down facing arrows to rearrange the order.

Usually you would put the highest quality codec first and then order them in descending quality.

When you double-click within a given softphone, you'll get something that looks like this.

The one on the left is an example of a Cisco call manager. Recall that you'll need a username, a password, and a proxy.

We put the authentication ID because in CUCM the digest credentials are right here.

And it's different than the extension that we're using within call manager.

You can also set a backup proxy so if the first one does not work it will try another one.

You can choose its transport, either UDP or TLS. Register with proxy, is usually yes.

But if you disable this, we won't register with a proxy.

If you have a customer that requires SIP trunking, then that’s probably what you're going to do.

And then you can see how different systems use Authentication ID.

Sometimes the username equals the authentication ID but for Cisco call manager it does not.

This is an example of a softphone using a hosted solution, where the domain was required by the provider.

And this is where you put the domain. It's optional.

It's not needed for call manager, it's not needed for a lot of different ones in fact,

but every once in a while you might need to use it.

And really the only way to know is just to ask them if it's required or not.

In this case, it was required to make those calls.

One more thing to note is the tone output.

If you go to the softphone properties, changing Tone Output to yes adds an additional PIN for your VOIP out block.

This will allow you to play back those tones within the room if you choose to do so.

Let’s talk about “Registration Timeout”. Within SIP, the Core will register not just once but over and over again.

There's a timeout in which you need to re register, even during the call, which is totally normal.

During long calls, you'll see re-registration happen several times.

If you didn't have that it would show it's registered forever even if the call server was no longer active.

Our Q-SYS Developers added this a while ago.

It's not needed for every case but it’s there just in case.

Alright gang, we're in the home stretch. We’re gonna review a few hot tips for VoIP troubleshooting when we get back.