Asterisk : Hard Phones | Types of Terminal Devices

The term hard in hard phones is the short form of hardware. Hardware phones are physical devices that act as a telephone handset. Hard phones are available for POTS (as used in the typical household) or VoIP. Hard phones will typically deliver the highest quality among types of terminal equipment. The most popular hard phones in the market today are:

• Grandstream GXP Series
  
• Linksys SPA Series
  
• Aastra 57 Series
  
• Cisco IP Phones (7940 & 7960)
  
• Polycom SoundPoint Series
  

Voice over IP uses various protocols depending on the handset, PBX, and the requirements. The major protocols supported by Asterisk are as follows.

H.323

The first protocol we will be looking at is H.323. Formally known as ITU-T Recommendation H.323, Packet-based multimedia communications systems, this is a suggestion on how to accomplish conferencing over IP, which includes voice, video, and data. This recommendation actually came at about the same time as SIP but has been more widely implemented.

The H.323 standard enjoys full backward compatibility. Currently H.323v5 is out, and v6 is being discussed. Each new release keeps all the pieces of the previous version. This gives a clear upgrade path and some assurance that the equipment won't be quickly antiquated.

H.323 equipment is widely available. From gateways to telephone handsets, all of the needed equipment is relatively easy to find. Most of the telephone handsets are full-featured because the H.323 protocol has a robust feature set.

While the H.323 standard was not designed for wide area networks, a whole set of rules allowing cross-domain addressing have been created. A system for reporting Quality of Service (QoS) back to a server has also been developed, allowing such information to be used to route future calls.

Finally, H.323 as a standard supports call intrusion. New endpoints can be added dynamically to any conference (that is, a call) at any time.

Asterisk support for H.323 is not built in. Instead, an additional package, asterisk-oh323, must be installed. After installation, H.323 handsets and gateways can be addressed much like any other channel in Asterisk.

SIP

The Session Initiation Protocol (SIP), is another method of signaling VoIP calls. SIP is a part of the default installation of Asterisk.

Most of the new VoIP equipment supports SIP. SIP has a number of advantages. One such advantage is that the code is smaller. The reason for this is that SIP only supports very basic features. All advanced features are supported through separate Internet standards. Another reason for its small footprint is that, as features are deprecated, the code to implement them is ousted.

Another advantage of SIP's design is its modular nature. As such, extending the protocol is easier to do. It also scales better and was designed with a large network in mind.

SIP seems to be the future of VoIP. There are many features that H.323 has, but are not available on SIP. This includes handset conference control, better Media Gateway definitions, and data sharing. However, SIP is a very good protocol for simple phone calls. Also, as we are using Asterisk, conferences are controlled by Asterisk, not the handsets. Asterisk is a clear Media Gateway, and when used as such, the ambiguity in SIP is not an issue.

IAX

The Inter-Asterisk eXchange (IAX) protocol is a protocol created by the programmers who brought us Asterisk. Due to the limitations of SIP and H.323, they chose to create a new de facto standard that would allow Asterisk servers to accomplish many things that are simply impossible with the other standards. They also support some features that are extremely difficult to do in SIP and H.323.

First, IAX pierces Network Address Translation (NAT) easily. Most firewalls and home Internet gateways use NAT, as well as some service providers. SIP and H.323 have worked hard to develop standards to allow them to break through the different types of NAT. However, IAX can work through most NAT devices right out of the box.

IAX is more configurable than the other protocols when dealing with Asterisk. As the source code is available, we can modify it if we so desire, and then submit those changes to be evaluated for inclusion in future versions of Asterisk. As IAX is not currently an Internet standard per se, there is no standard body to work through, allowing more rapid improvement and growth.

IAX supports the trunking of calls. This means that multiple calls can be combined through a single stream. Through the trunking capability, a significant amount of bandwidth can be saved by not having the overhead of multiple streams.

IAX connections between servers support the switch command with which information on how a call is routed can be efficiently shared between Asterisk servers.

IAX supports a large number of codecs. Any codec supported in Asterisk can be used with channels of this type.

As IAX is an Asterisk-created protocol, there are not many handsets and gateways available. However, as time passes, more and more devices are supporting the IAX protocol.

Just as a note, we sometimes see IAX and IAX2 differentiated. IAX2 has been merged into IAX, and IAX has been deprecated. Thus, if a device claims to support IAX2, it should really be supporting IAX.