Telecom Terms and Concepts
Pages: 1, 2
Packaging all the DS0s into a single physical connection has many advantages. The destination number can be passed over the signaling channel which means that a phone number isn't tied to a particular time slot, so many phone numbers can be used on a single access (these phone numbers are usually called Direct Inward Dials (DID)). Also, the number of DS0s you need is determined by how many simultaneous phone calls you want to have, not how many phones you have. Here lies another key difference between traditional and IP telephony: unused trunks are wasted in the traditional model, but they can be used for other things in the IP world.
While having a T1-type service to your large business might be feasible, 23 channels are a bit much for many smaller cases. There are two common solutions here, one is ISDN Basic Rate Interface (BRI), which provides two B channels and a D channel, along with the signaling available in the PRI service. The other option is to have a series of analog lines provided off an FXS port in the CO. Rather than plugging a phone into the line we provide a Foreign Exchange Office (FXO) port. A modem is an example of FXO, it can put the phone on and off hook, dial, and speak. Regular analog service, however, doesn't provide the signaling available in digital service, so giving individual DIDs to your employees isn't possible. The PBX answering the call can easily provide a menu that directs the caller to the appropriate internal extension, called an Auto Attendant. There is, however, an analog trunk service available which, when combined with different hardware, can provide the information necessary to have DID service.
When moving to the VoIP world things change significantly. The underlying transport, IP, is connectionless. This means that each packet is passed from hop to hop without the path being nailed up. The downside is that it's harder for us to allocate the network resources in advance, but it lets us make better use of the resources we have. The voice traffic itself is carried over Real Time Protocol (RTP), which itself runs over UDP. RTP adds sequencing information to the stream so that out of order packets can be detected. Note that RTP is still unreliable in that it doesn't retransmit missing or malformed packets, since a voice packet that arrives late or out of order is unhelpful.
Voice can be encoded in different ways which allows for compression. G.711 transmits the raw 64kbit/S PCM data in packets of 20msec. G.729 compresses the data down to 8Kbit/S with little loss in voice quality. There are many different codecs available, and both ends must agree on the same one to talk. Transcoding is the process whereby the voice stream is changed from one codec to another, which is used when the two endpoints can't use the same codec. Since voice compression is inherently lossy, multiple transcodings severely degrade the voice quality and this practice should be avoided.
Though the voice bearer traffic is easy enough to understand, it is the signaling that sets up and tears down calls that is complex. There are several protocols that can be used to do this, the most popular being H.323 and SIP.
H.323 is an older protocol (actually, a series of protocols) that was created by the International Telecommunications Union (ITU) for voice and video communications. SIP, the Session Initiation Protocol, is a creation of the IETF who also bring you such novel concepts as the internet itself. SIP was designed to carry a variety of information, from call setup, to presence and Instant Messaging. For either choice the purpose and operation are similar.
Consider the case of an IP telephone user who wants to call another user over the internet. Generally the phone itself has a network connection to a server that provides the call control, and for the sake of simplicity let's assume both phones use the same server. One phone dials a number (or a SIP address such as user@example.com) which is sent to the server.
The server checks to see if the call is allowed, and then invites the dialed party's phone to join the call. At the same time, a message is passed back to the caller's phone that the call is in progress, which results in the familiar ringback tone being played to the caller. When the dialed party's phone goes off hook, it sends a message to the server indicating it has been answered, which is again relayed back to the caller. This message is then acknowledged back to the server and then in turn to the phone.
At this point both phones are ready to talk using RTP. However, the flow of the packets is between the two phones--the server doesn't get involved at all. Unless transcoding is needed, or there are reasons to have the server interject, having the two phones call each other is the best use of resources. Removing the server from the flow also allows for better load balancing and reliability.
This model can be extended to multiple servers or even to the PSTN. In the latter case, one of the phones may actually be a gateway to a PRI. When the call gets to the gateway it dials out the PSTN to the remote telephone. There's nothing saying we can't do a lookup beforehand to see if the called party is better reached over the internet or the PSTN. The IP phone itself can also be an Analog Terminal Adapter (ATA) which connects a standard analog set to the IP network. If you've ever used a service like Vonage, this is what you're using. There's no limit to how we treat calls now that the intelligence has been moved closer to the user's hands.
The traditional PSTN is a complex creature, but it must be understood in order to integrate successfully with VoIP. Even though VoIP protocols borrow from traditional telephony concepts, the shift of intelligence from the core to the edge opens up a whole suite of new applications.
Sean Walberg is currently a Senior Network Engineer with a large Canadian financial services company, where he is building up the voice and data networks.
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