Phone Systems, VoIP, SIP?

PBX stands for Private Branch Exchange. Ezeetel PBX is our implementation of PBX - taken to a next level. Ezeetel PBX users share a number of outside lines for making external phone calls. Local calls, between the users, are available as well. On top of that, PBX offers many additional features, such as:

• Ring groups
• Conferences
• Interactive Voice Responses
• Queues
• Voicemail boxes
• Realtime monitoring
• Call recordings

and much more...

VoIP stands for Voice over Internet Protocol and it refers to the diffusion of voice traffic over internet-based networks. VoIP is widely used to cut the cost of traditional PSTN and some of its major benefits are:

More than one phone call on the same line.

Advance features, such as call forwarding, caller ID or automatic redialing, are simple and usually free.

Communications can be secured with encryption

SIP stands for Session Initiation Protocol. It is an IP telephony signalling protocol used to establish, modify and terminate VOIP telephone calls. SIP has taken the VOIP world by storm. The protocol resembles the HTTP protocol, is text-based, and very open and flexible. It has therefore largely replaced the H323 standard.

SDP stands for Session Description Protocol. It is a format for describing streaming media initialization parameters. Streaming media is content that is viewed or heard while it is being delivered.

Echo cancellation is the process of removing echo from a voice communication in order to improve voice call quality. Echo cancellation is often needed because speech compression techniques and packet processing delays generate an echo. There are 2 types of echo: acoustic echo and hybrid echo. Echo cancellation not only improves quality but also reduces bandwidth consumption because of its silence suppression technique.

RTP stands for Real Time Transport Protocol. It defines a standard packet format for delivering audio and video over the internet.

RTCP stands for Real-Time Transport Control Protocol. It works hand in hand with RTP. RTP does the delivery of the actual data, whereas RTCP is used to send control packets to participants in a call. The primary function is to provide feedback on the quality of service being provided by RTP.

A SIP URI is basically a user’s SIP phone number. The SIP URI resembles an e-mail address and is written in the following format: SIP URI = sip:x@y:Port, where x=Username and y=host (domain or IP) Examples:

sip:[email protected]

sip:[email protected]

sip:[email protected]

SIP uses Methods and Requests to establish a call session.

SIP Requests: INVITE = Establishes a session ACK = Confirms an INVITE request BYE = Ends a session CANCEL = Cancels establishing of a session REGISTER = Communicates user location (hostname, IP) OPTIONS = Communicates information about the capabilities of the calling and receiving SIP phones

SIP responses: 1xx = informational responses, such as 180, which means ringing 2xx = success responses 3xx = redirection responses 4xx = request failures 5xx = server errors 6xx = global failures

1xx = informational responses

100 Trying

180 Ringing

181 Call Is Being Forwarded

182 Queued

183 Session Progress

2xx = success responses

200 OK

202 accepted: Used for referrals

3xx = redirection responses

300 Multiple Choices

301 Moved Permanently

302 Moved Temporarily

305 Use Proxy

380 Alternative Service

4xx = request failures

400 Bad Request

401 Unauthorized: Used only by registrars. Proxys should use proxy authorization 407

402 Payment Required (Reserved for future use)

403 Forbidden

404 Not Found: User not found

405 Method Not Allowed

406 Not Acceptable

407 Proxy Authentication Required

408 Request Timeout: Couldn't find the user in time

410 Gone: The user existed once, but is not available here any more.

413 Request Entity Too Large

414 Request-URI Too Long

415 Unsupported Media Type

416 Unsupported URI Scheme

420 Bad Extension: Bad SIP Protocol Extension used, not understood by the server

421 Extension Required

423 Interval Too Brief

480 Temporarily Unavailable

481 Call/Transaction Does Not Exist

482 Loop Detected

483 Too Many Hops

484 Address Incomplete

485 Ambiguous

486 Busy Here

487 Request Terminated

488 Not Acceptable Here

491 Request Pending

493 Undecipherable: Could not decrypt S/MIME body part

5xx = server errors

500 Server Internal Error

501 Not Implemented: The SIP request method is not implemented here

502 Bad Gateway

503 Service Unavailable

504 Server Time-out

505 Version Not Supported: The server does not support this version of the SIP protocol

513 Message Too Large

6xx = global failures

600 Busy Everywhere

603 Decline

604 Does Not Exist Anywhere

606 Not Acceptable

Example of SIP Call session between 2 phones

A sip call session between 2 phones is established as follows:

The calling phone sends out an invite

The called phone sends an information response 100 – Trying – back.

When the called phone starts ringing a response 180 – Ringing – is sent back

When the caller picks up the phone, the called phone sends a response 200 – OK

The calling phone responds with ACK – acknowledgement

Now the actual conversation is transmitted as data via RTP

When the person calling hangs up, a BYE request is sent to the calling phone

The calling phone responds with a 200 – OK.

FOIP stands for Fax over IP. It refers to the process of sending and receiving faxes over a VOIP network. Fax over IP works via T38 and requires a T38 capable VOIP gateway as well as a T38 capable fax machine, fax card or fax software. PBXware includes a compatible T38 fax service.

DID stands for Direct Inward Dialing (also called DDI in Europe). It is a feature used with PBX systems, whereby the telephone company allocates a range of numbers associated with one or more phone lines. The purpose of DID is to allow a company to assign a personal number to each employee, without requiring a separate phone line for each.

VoIP system requires the use of VoIP phones. These come in several versions/types:

VoIP Softphones

USB VOIP phones

Hardware SIP Phone

Analog phone via an ATA adapter

NOTE: ATA adapter allows an analog phone to be connected to a VoIP system

FXS and FXO are the names of ports used by Analog phone lines.

FXS - Foreign eXchange Subscriber is a port that delivers the analog line to the subscriber.

FXO - Foreign eXchange Office is a port that receives the analog line. Since the FXO port is attached to a device, such as a fax or a phone, the device is often called the ‘FXO device’.

FXO and FXS are always paired, similar to a male/female plug.

A SIP server is the main component of an IP PBX, dealing with the setup of all SIP calls in the network. A SIP server is also referred to as a SIP Proxy or a Registrar. An example of a SIP server is our Ezeetel PBX