Voip dtmf

Question

Hello,
I'm trying to understand how dtmf is different from analog world to the VOIP world.  In the analog world from my understanding dtmf are tones frequency so when caller dials numbers then does numbers have different tones frequency and it send to the telephone company to decode those tone frequencies and know where to route that call to the other person receiving the call. In the VOIP world with call manager there is in band or out band dtmf. But how does the process work when phone A calls phone B how does it know where to route the call. Like does dtmf turn into packets to tell cucm where to route the call and the user hears tones when pressing the numbers just to simulate the dtfm in analog world. What about dtmf in band with rtp packets does dtfm go first in rtp to let call be routed to the other phone  or those the voice gateway or cisco switch see rtp dtmf and route to the correct phone?
Thanks,

Answer 1

Horacio,
What you are describing is call setup or call signaling. In particular, you are asking about how the client (IP Phone) communicates its intended destination to the server (CUCM).
In a VoIP environment, the client sends the dialed digits to the server using IP packets. Each call signaling protocol has a method for communicating this information in what is called an application header. The information does not rely on DTMF tones. The client will simply identify the digit(s) dialed in the appropriate header.
There are two general ways that a client, such as an IP phone, can present the dialed digits. One way is called "en bloc" or "all together". What this means is that the phone will send a packet to the server and the call setup request will provide all digits as a single "string". An example is your mobile phone. You can dial all of the digits of your intended party and then click "send" or "call". When you do this, your phone is sending all of the dialed digits to your carrier. This is en bloc dialing.
In CUCM, phones use en bloc dialing in several scenarios. Including redial and dialing from the corporate directory (or missed calls, received calls, etc.).
The second way a client can communicate dialed digits is digit-by-digit. CUCM supports digit-by-digit dialing with Cisco IP Phones that support SCCP and SIP. All this means, is as you dial a digit on your phone, that phone is sending a packet to the CUCM that identifies you dialed that single digit. The CUCM digit analysis process is collecting those digits, one-by-one. As soon as it determines there is a unique match, it will route the call.
None of what I described above leverages DTMF. Meaning, there are no tones exchanged. There is no need. The client just says "this dude dialed a 5", or whatever you dialed.
Now, you mentioned voice gateways. Well, a voice gateway that communicates with the CUCM is leveraging an IP protocol. In today's networks the protocol is usually SIP, H.323, or MGCP. Of course, Cisco gateways can also use SCCP, which is proprietary to Cisco. Regardless of the protocol used, the voice gateway sends digits to the CUCM and receives digits from the CUCM in a manner that is similar to IP phone clients. Which is to say, that they stick the destination information in a header, stick that header in a packet, and send it to the appropriate peer. No DTMF.
Of course gateways, by there very nature, connect two disparate systems. Gateways relay call setup information from one entity to another. For instance, let's say you have a T1 PRI connected to a voice gateway. The gateway has an IP connection to CUCM and an ISDN connection to the carrier (or whatever is on the other end of the PRI). It just so happens that the ISDN protocol also exchanges call setup information in a manner similar to IP protocols. Which is to say that the digits dialed by the calling party are exchanged in protocol messages NOT DTMF tones.
Up to this point, I have only focused on call setup because that seemed to be the premise of your question. I am not suggesting that DTMF isn't used in VoIP. It is. DTMF is used by applications such as voicemail systems, call centers, and other IVR-based systems. For example, let's say you call a given number and you hear a greeting which prompts you to press 1 to connect to Horacio and press 2 to connect to bill. This is an IVR system and how that system does its job is by interpreting the key presses using DTMF recognition.
This gets back to another thread you posted concerning "in band" and "out of band" (OOB) DTMF. "In band" simply means that the DTMF tones are packetized and sent in the RTP stream. They are digital samples of the analog tone, literally. The other end is responsible for understanding how to deal with that. If you and I were on a phone call and I kept hitting the number 5, you would hear it and you'd probably get bent because your ear isn't equipped to understand the DTMF representation of the digit "5".
Out of band (OOB) means that the sender is relaying the digits dialed via the call signaling protocol. It is similar to the whole digit-by-digit thing I described earlier but it is presented in a different message. This message gets relayed through your network, and any intermediary devices, and lands on the receiver's end. As long as every device in the call flow is using the same method to do the OOB signaling, you are golden. This part of the conversation can get long. Longer than it already has. I recommend researching "DTMF Relay" to start getting a better understanding.
HTH
-Bill
(b) http://ucguerrilla.com
(t) @ucguerrilla
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