VOIP Telephony Concepts used with Exchange Unified Messaging

VOIP Telephony Concepts used with Exchange Unified Messaging

To successfully deploy Unified Messaging in an Organization, you need to know Voice mail and telephony concepts such as Circuit Switching, Packet Switching and VOIP.

Circuit Switched Network

It is a dedicated connection that connects two nodes so that the two nodes can communicate with each other. After a call is established only two nodes can use the connection until one of the two parties ends the call. Circuit Switched Network ensures a level of service by transferring large amount of data with guaranteed transmission capacity. There are two types of Circuit Switched network, they are:
· Analog Circuit Switched Network
· Digital Circuit Switched Network

Public Switched Telephony Network (PSTN)

PSTN is a technology using which multiple calls can be transmitted over the same physical medium. PSTN shares available bandwidth more efficiently over the same physical network connectivity.

Connectivity Concepts

T1 and E1 Lines: They work more or less in the similar war and can carry multiple digital voice channels. T1 is used mostly in North America and Japan while E1 is mostly used at UK, Australia and New Zealand. T1 can carry 24 channels per frame, out of which 23 are used for voice while one is used for data signaling. E1 can carry up to 32 channels per frame out of which 30 are used for voice and 2 are used for data signaling.

Trunk Lines: Trunk lines are used to connect Corporate PBX to the Phone Company’s Central Office Switch.

Tie Lines: Tie Lines are T1 or E1 lines that is used to connect 2 or more Corporate PBX systems using Inter-PBX Network signaling or Protocols.

Time Division Multiplexing (TDM): TDM is a technology for transmitting a number of separate voice signals separately over a single physical high Bandwidth phone line. Using this technique the line can be divided into smaller multiple fixed bandwidth channels, each carrying its own voice signal. This shares the transmission productivity of the high bandwidth line. This is how T1 and E1 lines are divided.

Analog PBX systems

Analog PBX systems use the following protocol

In-Band Dual Tone Multi Frequency (DTMF) signaling: It defines a Protocol where call signaling contains DTMF tones. These are within the voice frequency range and are carried in the same channel as voice. The Call Diversion information is supplemental signaling, but it is required to support voice mail.

Out of Band RS232 Signaling: It defines a supplementary protocol where the call signaling is carried on a separate channel from voice. This is done on a separate wire or serial connection using RS232. Out of Band RS 232 signaling is also known as Simplified Message Desk Interface (SMDI).
Exchange Unified Messaging uses VOIP Gateway to transfer analog circuit switching protocols to packet switching protocols. VOIP gateway interprets and translates the call diversion information from the PBX specific protocol to Unified Messaging supported SIP protocol.

Digital PBX systems

Digital PBX system uses the following protocols

Set Emulation: it is used when the signaling protocol is proprietary to the PBX vendor. In this case VOIP gateway must be able to interpret the protocol and translate to Unified Messaging understandable protocol.

QSIG: it is a signaling protocol that is based on ISDN Q.931 standard. QSIG is used between corporate PBX systems and it allows multiple PBX systems to operate together in feature transparent way. Therefore large and distributed Organizations can appear to have a single phone system, though they may have multiple PBX system. QSIG can also be used as Circuit Switch and signaling protocol translated by VOIP gateway.

Channel Associated Signaling (CAS): It is a signaling protocol associated with each channel of voice in a T1 environment. Within the channel, bits are robbed and replaced with basic call signaling information. However, basic CAS does not include the signaling requirements for voice mail. These are provided through In-Band DTMF or RS232 signaling.

Voice over IP (VOIP): It is a technology where voice data is sent in packets by using IP instead of traditional circuit transmission or circuit switched telephone lines of PSTN.

VOIP protocols used by Unified Messaging

SIP: It is a text based application layer signaling and a call control protocol. It is used for initiating, modifying and ending an interactive user session, which involves multimedia elements such as voice, video, instant messaging, online games and virtual reality. It supports both unicast and multicast communication. SIP is used only for setting up and tearing down voice or video calls. SIP uses RTP for transferring digitized audio data between parties participating in a call. Each RTP packet contains one or more media pay loads and other relevant information, such as time stamp and sequence numbers.

Real-time Transport Protocol (RTP): It is standard packet format for delivering audio and video over a given network. RTP uses dynamic ports that are negotiated between sender and receiver.

T.38: It is a protocol that allows you to send FAX messages over an IP based network. Then the IP based network uses SMTP and MIME to send the message to the recipient’s mailbox.

Call Diversion Information

While using Unified Messaging systems you need called party and call diversion information so that the system is aware of the recipients. When an outside caller dials a number of someone inside the business, the call is routed to the Central Office and then to the appropriate customer site’s PBX system. The PBX routes the call to the appropriate desk phone. If the recipient of the call is not at the desk, the call is directed to Ring No Answer. Then, the PBX uses it’s call coverage information to check where the unanswered call should be routed.

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