Video Transcript
Intro to VoIP Telephony (Part 3)
6m 43s
00:07
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.
00:16
DTMF is used for sending numbers and issuing commands when dialing.
00:21
For example, 697 Hz and 1209 Hz are the two tones that make up the number one.
00:29
There are two types of tones, there's in-band and out-of-band.
00:33
So when we're talking about in-band we're talking about RTP streams,
00:38
and out-of-band means that the digits are sent in SIP messages instead of RTP.
00:43
There are two versions that we support within the Q-SYS softphone: RFC 2833 and DTMF info.
00:52
RFC 2833 is the SIP standard sending DTMF, while DTMF INFO is old and not really used anymore.
01:00
DTMF in-band shares the same RTP streams with the DTMF tones as voice. It's often heard by all participants.
01:08
So if a customer says, “hey I don't really want to hear all the button presses and all that stuff,”
01:13
well you can't really use DTMF in-band.
01:16
Now there is no formal in-band DTMF tones built into Q-SYS but you can generate your own DTMF tones
01:23
you can embed into your own audio traffic if you need to.
01:26
Just build the tones into a sine generator and expose the proper pins to trigger them.
01:31
Then you'd mix the DTMF into your microphones, and then send them to your Q-SYS VoIP out component.
01:38
And then you would use your VoIP IN component block to receive the far end audio
01:42
and send those to the loudspeakers.
01:44
The VOIP status in the control block is relatively simple,
01:47
which acts as a tone controller for all these different tones.
01:51
In this example we've got the DTMF playback enabled and we've got a ringtone enabled.
01:56
We also enabled entry and exit tones. These are the tones that you do that with.
02:02
If you notice, this is going to be a single tone because it's a single file.
02:07
This is because if you put an actual DTMF tone in there there's a chance that you'd get a double detection.
02:13
If you're actually hitting the buttons and it's playing back on the loudspeaker
02:17
you could be detecting that as a DTMF in your own system as well.
02:22
Not really what you want, right?
02:24
It makes more sense to just use a different tone that you know is not going to match a DTMF tone.
02:30
This is the status and control block, where you’d push all the DTMF buttons while you're in the call.
02:35
Beyond the basics, we do have the do not disturb, connect, disconnect, off hook, ringing, and turn on auto answer.
02:46
You’ll also see the status of the softphone, which is communicated by the SIP.
02:51
We also have this option for “continual DTMF” and “simulate incoming calls” which are for troubleshooting.
02:58
This screen is the softphone Configuration screen from Core Manager.
03:02
At the top, you can specify which LAN you want the Core to use for VoIP.
03:07
Generally speaking, most installs will use LAN A for Q-LAN audio
03:12
and use a LAN B to connect to your enterprise connection and VoIP.
03:16
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.
03:24
5060 is the standard port for SIP so many times you don't have to change that.
03:29
Different solutions may require you to use a different port.
03:32
Like Ring Central, for example, uses 5090.
03:37
Below that, you can enable DTMF info, which is set to “no” by default,
03:42
but you would need to enable it if you are using an Avaya system.
03:47
Next is RTP Type which is usually 101.
03:50
When you set your Core to Stun, you’ll receive immediate praise from James Tiberious Kirk himself,
03:55
and also that sentence is utter nonsense.
03:58
Enable Stun to traverse NAT if you are behind a firewall.
04:03
So if you have a hard time reaching outside the network or the enterprise firewall, you can always turn that on.
04:09
Generally you would “Enable SRTP” traffic if it's a government client or something like that.
04:15
Here’s where you're going to enable your preferred audio codec
04:18
and the order that you want them to default to.
04:22
Use the up and down facing arrows to rearrange the order.
04:25
Usually you would put the highest quality codec first and then order them in descending quality.
04:31
When you double-click within a given softphone, you'll get something that looks like this.
04:36
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.
04:44
We put the authentication ID because in CUCM the digest credentials are right here.
04:49
And it's different than the extension that we're using within call manager.
04:53
You can also set a backup proxy so if the first one does not work it will try another one.
04:57
You can choose its transport, either UDP or TLS. Register with proxy, is usually yes.
05:03
But if you disable this, we won't register with a proxy.
05:06
If you have a customer that requires SIP trunking, then that’s probably what you're going to do.
05:11
And then you can see how different systems use Authentication ID.
05:15
Sometimes the username equals the authentication ID but for Cisco call manager it does not.
05:21
This is an example of a softphone using a hosted solution, where the domain was required by the provider.
05:27
And this is where you put the domain. It's optional.
05:31
It's not needed for call manager, it's not needed for a lot of different ones in fact,
05:35
but every once in a while you might need to use it.
05:38
And really the only way to know is just to ask them if it's required or not.
05:42
In this case, it was required to make those calls.
05:45
One more thing to note is the tone output.
05:48
If you go to the softphone properties, changing Tone Output to yes adds an additional PIN for your VOIP out block.
05:56
This will allow you to play back those tones within the room if you choose to do so.
06:01
Let’s talk about “Registration Timeout”. Within SIP, the Core will register not just once but over and over again.
06:10
There's a timeout in which you need to re register, even during the call, which is totally normal.
06:15
During long calls, you'll see re-registration happen several times.
06:19
If you didn't have that it would show it's registered forever even if the call server was no longer active.
06:26
Our Q-SYS Developers added this a while ago.
06:29
It's not needed for every case but it’s there just in case.
06:33
Alright gang, we're in the home stretch. We’re gonna review a few hot tips for VoIP troubleshooting when we get back.
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Intro to VoIP Telephony (Part 3)
6m 43s