Voice over IP Online Course
Lesson 3 - How Does VoIP Work?
B. Call Manager
Call Manager
- VoIP needs a "call manager" to facilitate calls
- Direct node to node IP calls are rare
- Must map telephone numbers or names to IP addresses
- Standalone dedicated box or built into routers
- Uses either standard protocols H.323 or SIP, or proprietary, such as Cisco's SKINNY
The call manager on an IP Telephone system provides the management of the telephone calls, similar to the function that SS7 does for the PSTN. Although in theory two IP telephones could communicate directly, this would require that one user knows the IP address of the other user's telephone, an unlikely occurrence. The call manager is also called a gatekeeper. The call manager provides some or all of the following services.
Call establishment
Establishing a call requires address resolution. This means that the way that the originating caller identifies the recipient, by telephone number, name, extension or IP address must be mapped to that recipient's IP address if the recipient is also using Voice over IP. On the other hand, if the recipient is on an external telephone system, the call manager must forward the request to the PSTN access device, the VoIP gateway.Call initiation
The call manager then proceeds to initiate the call, by using H.225, SIP, or proprietary signaling to contact the called party. Upon a successful connection, the gatekeeper hands off the call to the two connected phones or phone-gateway pair. If the call cannot go through, because the called phone was in use or otherwise unavailable, the gatekeeper will then inform the caller with a busy signal or redirect the call to an automated attendant or voice mail system.Admissions control
The call manager can control access to the telephone system to authorized and registered endpoints. Devices unknown to the administrator will be disallowed.Bandwidth control
Connections can be disallowed if the system has no more bandwidth. A percentage of the bandwidth of the network may be reserved for data or some critical usage. The call manager may also restrict the number of people participating in a video/audio conference.Zone management
The call manager will manage its zone, defined as the endpoint devices that register with it. This last includes maintaining a real time list of calls in progress in order to provide a busy signal as required.Additional services
The call manager provides some of the services that a traditional PBX provides such as call hold, call transfer, call forwarding, and call waiting.The call manager can be a dedicated box, be integrated into another box such as a router or be software sitting on a server.
The call control functions used by the call manager are H.323 and SIP plus some proprietary protocols such as Cisco's SKINNY protocol.
Figure 10: Setting up a call
As figure 10 illustrates, when the phone at one end is picked up a connection is made to the call manager which in turn contacts the second VoIP device. Only when the call setup is completed do the two endpoint devices communicate with each other on a peer-to-peer basis.
Call Setup Services
- H.323
- Session Initialization Protocol (SIP)
H.323
H.323 is a set of protocols used by VoIP to establish and manage telephone calls. H.323 is controlled by the International Telephone Union (ITU).Not only voice
H.323 is a very broad set of standards that provide for video and data conferencing as well
as audio. The networks that H.323 was designed to work with do not provide quality of service.
Ethernet is the primary example of this type of network. H.323 is not specific to Ethernet
or any network; it will work with them all. And if the network does have some QoS, bonus.
Figure 11: The H.323 protocol suite
Many protocols
An umbrellaH.323 is actually a framework for multimedia conferencing and therefore uses other protocols to handle many functions. See figure 11.
- Control and call signals
H.245 used to negotiate channel usage and capabilities
Q.931 used for call signaling and call setup
RAS Registration/Admission/Status is a protocol used to communicate with a Gatekeeper - Audio codecs G.711, G.722, G.723, G.728, G.729
- Video codecs H.261, H.263
- RTP/RTCP - used for sequencing audio and video packets
H.323 components
H.323 defines four major components for a network-based communications system: Terminals, Gateways, Gatekeepers, and Multipoint Control Units.- Terminals on a VoIP system are telephones and PCs equipped with VoIP capabilities.
- Gateways are the bridge to the PSTN.
- Gatekeepers act as the central point for all calls within its zone and provides call control services to registered endpoints. In a previous discussion, the H.323 Gatekeeper was called the Call Manager. To recap, the functions of the Gatekeeper include address resolution and mapping, call initiation and establishment, admissions control, bandwidth control and zone management.
- Multipoint Control Unit (MCU) supports conferences between three or more endpoints. Conferencing can be audio, video or data.
Figure 12: H.323 zones
First on the scene, but
H.323 is a flexible and comprehensive framework for multimedia conferencing, including VoIP, which has been implemented on millions of end devices since 1996 (V1) and 1998 (V2). It is known to work well and is reliable. However, there is room for an alternative call management service because of the following features of H.323.- H.323 is not dedicated to VoIP and therefore is complicated because of the extra baggage it carries around for video conferencing.
- H.323 is not very flexible. New features can be added but they must be backward compatible.
- H.323 is controlled by the ITU, which moves slowly with new technology, and is slanted toward the traditional PSTN. In the fast paced world of the Internet, H.323 is evolving slowly.
SIP
SIP, the Session Initiation Protocol, is a signaling protocol for Internet conferencing, telephony, presence, events notification and instant messaging. SIP was developed by the Internet Engineering Task Force (IETF).SIP is used for setting up, controlling and tearing down sessions on the Internet. Sessions include, but are not limited to, Internet telephone calls and multimedia conferences. SIP is also used for instant messaging and presence. Note that SIP is designed for managing sessions (connections) whereas H.323 was designed for multimedia conferencing. VoIP falls within the scope of both.
SIP is a request-response protocol that closely resembles two other Internet protocols, HTTP and SMTP (the protocols that power the world wide web and email); consequently, SIP sits comfortably alongside Internet applications. Using SIP, telephony becomes another web application and integrates easily into other Internet services.
Figure 13: SIP in action
SIP architecture
SIP uses the following components:SIP URL Every endpoint on the VoIP system has a SIP URL for identification. The URL for
Bob Jones at ABC.com might be:
sip:bobj@abc.com
If calling a telephone number on the PSTN, the URL will look like: SIP:5551212@gateway, where gateway is the name of the machine that acts as the gateway to the PSTN.
Registration server The registration server authenticates the user, and adds the mapping between URL and network address to the location server's database. When the user agent starts up, the first message it sends is a REGISTRATION.Location database The location database maintains the database of name to location (IP address usually) mappings. The information in the database is usually acquired from the user agent registrations, but may be acquired in other ways as well, such as DNS. The database may be queried in various ways, although LDAP is the most common.
When a user agent wants to connect to a remote SIP endpoint, it queries the location database in the location server for the contact information.Proxy server Proxy servers, as their name suggests, act on behalf of user agents, routing SIP messages to correct destinations.
Redirect server A redirect server differs from a proxy server in that it does not forward messages but simply does a location look-up and returns one (or more) addresses for the destination and leaves it up to the original user agent to contact the destination at these addresses directly.A SIP server will include some or all of the above functions or the functions can be split between multiple machines.
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