VoIP

How VoIP works

VoIP Benefits

FAQ about VoIP (Source: en.wikipedia.org)

Q. What is VoIP?
A. VoIP stands for Voice over IP or simply Voice over Internet. It's the new technology for transmitting voice over Internet. Other terms frequently used and synonymous with VOIP are IP telephony, Internet telephony, etc.

Q. What is Intranet?
A. An intranet is a private computer network that uses Internet Protocol technologies to securely share any part of an organization's information or network operating system within that organization. Or it's the Internet inside a company. The term is used in contrast to internet, a network between organizations, and instead refers to a network within an organization. Sometimes the term refers only to the organization's internal website, but may be a more extensive part of the organization's information technology infrastructure. It may host multiple private websites and constitute an important component and focal point of internal communication and collaboration.

Q. What is VLAN?
A. A virtual LAN, commonly known as a VLAN, is a group of hosts with a common set of requirements that communicate as if they were attached to the same broadcast domain, regardless of their physical location. A VLAN has the same attributes as a physical LAN, but it allows for end stations to be grouped together even if they are not located on the same network switch. Network reconfiguration can be done through software instead of physically relocating devices.

Q. What is QoS?
A. QoS stands for Quality of Service. It tells how good is your voice quality when you transmit voice over Internet. This is the main technical issue of VoIP. After many years of improvement, today's VoIP QoS is acceptable.

Q. What is SIP?
A. The Session Initiation Protocol (SIP) is an IETF-defined signaling protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP). The protocol can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions consisting of one or several media streams. The modification can involve changing addresses or ports, inviting more participants, adding or deleting media streams, etc... Other feasible application examples include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer and online games.

SIP was originally designed by Henning Schulzrinne and Mark Handley starting in 1996. The latest version of the specification is RFC 3261[1] from the IETF Network Working Group.[2] In November 2000, SIP was accepted as a 3GPP signaling protocol and permanent element of the IP Multimedia Subsystem (IMS) architecture for IP-based streaming multimedia services in cellular systems.

The SIP protocol is an Application Layer protocol designed to be independent of the underlying transport layer; it can run on Transmission Control Protocol (TCP), User Datagram Protocol (UDP), or Stream Control Transmission Protocol (SCTP).[3] It is a text-based protocol, incorporating many elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP),[4] allowing for direct inspection by administrators.

Protocol design

SIP employs design elements similar to the HTTP request/response transaction model.[5] Each transaction consists of a client request that invokes a particular method or function on the server and at least one response. SIP reuses most of the header fields, encoding rules and status codes of HTTP, providing a readable text-based format.

SIP works in concert with several other protocols and is only involved in the signaling portion of a communication session. SIP clients typically use TCP or UDP on port numbers 5060 and/or 5061 to connect to SIP servers and other SIP endpoints. Port 5060 is commonly used for non-encrypted signaling traffic whereas port 5061 is typically used for traffic encrypted with Transport Layer Security (TLS). SIP is primarily used in setting up and tearing down voice or video calls. It has also found applications in messaging applications, such as instant messaging, and event subscription and notification. There are a large number of SIP-related Internet Engineering Task Force (IETF) documents (Request for Comments) that define behavior for such applications. The voice and video stream communications in SIP applications are carried over another application protocol, the Real-time Transport Protocol (RTP). Parameters (port numbers, protocols, codecs) for these media streams are defined and negotiated using the Session Description Protocol (SDP) which is transported in the SIP packet body.

A motivating goal for SIP was to provide a signaling and call setup protocol for IP-based communications that can support a superset of the call processing functions and features present in the public switched telephone network (PSTN). SIP by itself does not define these features; rather, its focus is call-setup and signaling. However, it was designed to enable the construction of functionalities of network elements designated proxy servers and user agents. These are features that permit familiar telephone-like operations: dialing a number, causing a phone to ring, hearing ringback tones or a busy signal. Implementation and terminology are different in the SIP world but to the end-user, the behavior is similar.

SIP-enabled telephony networks can also implement many of the more advanced call processing features present in Signaling System 7 (SS7), though the two protocols themselves are very different. SS7 is a centralized protocol, characterized by a complex central network architecture and dumb endpoints (traditional telephone handsets). SIP is a peer-to-peer protocol, thus it requires only a simple (and thus scalable) core network with intelligence distributed to the network edge, embedded in endpoints (terminating devices built in either hardware or software). SIP features are implemented in the communicating endpoints (i.e. at the edge of the network) contrary to traditional SS7 features, which are implemented in the network.

Although several other VoIP signaling protocols exist, SIP is distinguished by its proponents for having roots in the IP community rather than the telecommunications industry. SIP has been standardized and governed primarily by the IETF, while other protocols, such as H.323, have traditionally been associated with the International Telecommunication Union (ITU).

The first proposed standard version (SIP 2.0) was defined by RFC 2543. This version of the protocol was further refined and clarified in RFC 3261, although some implementations are still relying on the older definitions.
[edit] SIP network elements

A SIP user agent (UA) is a logical network end-point used to create or receive SIP messages and thereby manage a SIP session. A SIP UA can perform the role of a User Agent Client (UAC), which sends SIP requests, and the User Agent Server (UAS), which receives the requests and returns a SIP response. These roles of UAC and UAS only last for the duration of a SIP transaction.[1]

A SIP phone is a SIP user agent that provides the traditional call functions of a telephone, such as dial, answer, reject, hold/unhold, and call transfer.[6][7] SIP phones may be implemented by dedicated hardware controlled by the phone application directly or through an embedded operating system (hardware SIP phone) or as a softphone, a software application that is installed on a personal computer or a mobile device, e.g., a personal digital assistant (PDA) or cell phone with IP connectivity. Examples include softphones such as Ekiga, KPhone, Twinkle, Windows Live Messenger, X-Lite, and hardware phones from vendors such as Avaya, Cisco, Leadtek, Nortel, Polycom, and Snom. As vendors increasingly implement SIP as a standard telephony platform, often driven by 4G efforts, the distinction between hardware-based and software-based SIP phones is being blurred and SIP elements are implemented in the basic firmware functions of many IP-capable devices. Examples are devices from Nokia and Research in Motion.

Each resource of a SIP network, such as a User Agent or a voicemail box, is identified by a Uniform Resource Identifier (URI), based on the general standard syntax[8] also used in Web services and e-mail. A typical SIP URI is of the form: sip:username:password@host:port. The URI scheme used for SIP is sip:. If secure transmission is required, the scheme sips: is used and SIP messages must be transported over Transport Layer Security (TLS).[1]

In SIP, as in HTTP, the User Agent may identify itself using a message header field 'User-Agent', containing a text description of the software/hardware/product involved. The User-Agent field is sent in request messages, which means that the receiving SIP server can see this information. SIP network elements sometimes store this information[9], and it can be useful in diagnosing SIP compatibility problems.

SIP also defines server network elements. Although two SIP endpoints can communicate without any intervening SIP infrastructure, which is why the protocol is described as peer-to-peer, this approach is often impractical for a public service.

RFC 3261 defines these server elements:

A proxy server " is an intermediary entity that acts as both a server and a client for the purpose of making requests on behalf of other clients. A proxy server primarily plays the role of routing, which means its job is to ensure that a request is sent to another entity "closer" to the targeted user. Proxies are also useful for enforcing policy (for example, making sure a user is allowed to make a call). A proxy interprets, and, if necessary, rewrites specific parts of a request message before forwarding it."

"A registrar is a server that accepts REGISTER requests and places the information it receives in those requests into the location service for the domain it handles."

"A redirect server is a user agent server that generates 3xx responses to requests it receives, directing the client to contact an alternate set of URIs.The redirect server allows SIP Proxy Servers to direct SIP session invitations to external domains."

The RFC specifies: "It is an important concept that the distinction between types of SIP servers is logical, not physical."

Other SIP related network elements are

Session border controllers (SBC), they serve as middle boxes between UA and SIP server for various types of functions, including network topology hiding, and assistance in NAT traversal.
Various types of gateways at the edge between a SIP network and other networks (as a phone network)

SIP messages

SIP is a text-based protocol with syntax similar to that of HTTP. There are two different types of SIP messages: requests and responses. The first line of a request has a method, defining the nature of the request, and a Request-URI, indicating where the request should be sent.[10] The first line of a response has a response code.

For SIP requests, RFC 3261 defines the following methods:[11]

* REGISTER: Used by a UA to notify its current IP address and the URLs for which it would like to receive calls.
* INVITE: Used to establish a media session between user agents.
* ACK: Confirms reliable message exchanges.
* CANCEL: Terminates a pending request.
* BYE: Terminates a session between two users in a conference.
* OPTIONS: Requests information about the capabilities of a caller, without setting up a call.

The SIP response types defined in RFC 3261 fall in one of the following categories:[12]

* Provisional (1xx): Request received and being processed.
* Success (2xx): The action was successfully received, understood, and accepted.
* Redirection (3xx): Further action needs to be taken (typically by sender) to complete the request.
* Client Error (4xx): The request contains bad syntax or cannot be fulfilled at the server.
* Server Error (5xx): The server failed to fulfill an apparently valid request.
* Global Failure (6xx): The request cannot be fulfilled at any server.

[edit] Instant messaging and presence

The Session Initiation Protocol for Instant Messaging and Presence Leveraging Extensions (SIMPLE) is the SIP-based suite of standards for instant messaging and presence information. During an instant message session, files can be transferred using, for example, MSRP (Message Session Relay Protocol).

Some efforts have been made to integrate SIP-based VoIP with the XMPP specification. Most notably Google Talk, which extends XMPP to support voice, plans to integrate SIP. Google's XMPP extension is called Jingle and, like SIP, it acts as a Session Description Protocol carrier.
[edit] Conformance testing

TTCN-3 test specification language is used for the purposes of specifying conformance tests for SIP implementations. SIP test suite is developed by a Specialist Task Force at ETSI (STF 196).[13]
[edit] Applications

Many VoIP phone companies allow customers to bring their own SIP devices, as SIP-capable telephone sets, or softphones. The market for consumer SIP devices continues to expand, there are many devices such as SIP Terminal Adapters, SIP Gateways etc.

The free software community started to provide more and more of the SIP technology required to build both end points as well as proxy and registrar servers leading to a commodification of the technology, which accelerates global adoption. As an example, the open source community at SIPfoundry actively develops a variety of SIP stacks, client applications and SDKs, in addition to entire private branch exchange (IP PBX) solutions that compete in the market against mostly proprietary IP PBX implementations from established vendors.

The National Institute of Standards and Technology (NIST), Advanced Networking Technologies Division provides a public domain implementation of the JAVA Standard for SIP JAIN-SIP which serves as a reference implementation for the standard. The stack can work in proxy server or user agent scenarios and has been used in numerous commercial and research projects. It supports RFC 3261 in full and a number of extension RFCs including RFC 3265 (Subscribe / Notify) and RFC 3262 (Provisional Reliable Responses) etc.
[edit] SIP-ISUP interworking

SIP-I, or the Session Initiation Protocol with encapsulated ISUP, is a protocol used to create, modify, and terminate communication sessions based on ISUP using SIP and IP networks. Services using SIP-I include voice, video telephony, fax and data. SIP-I and SIP-T[14] are two protocols with similar features, notably to allow ISUP messages to be transported over SIP networks. This preserves all of the detail available in the ISUP header, which is important as there are many country-specific variants of ISUP that have been implemented over the last 30 years, and it is not always possible to express all of the same detail using a native SIP message. SIP-I was defined by the ITU-T, where SIP-T was defined via the IETF RFC route.[15]

Q. What is IAX?
A. IAX is the Inter-Asterisk eXchange protocol native to Asterisk PBX and supported by a number of other softswitches and PBXs. It is used for enabling VoIP connections between servers beside client–server communication.

IAX now most commonly refers to IAX2, the second version of the IAX protocol. The original IAX protocol has been deprecated in favor of IAX2.

The IAX2 protocol was published as an informational (non-standards-track) RFC 5456 by discretion of the RFC Editor in February 2010.[1]

IAX2 is a VoIP protocol that carries both signaling and media on the same port. The commands and parameters are sent in binary format and any extension has to have a new numeric code allocated. Historically this was modeled after the internal data passing of Asterisk modules[citation needed].

IAX2 uses a single UDP data stream (usually on port 4569) to communicate between endpoints, multiplexing signaling and media flow. IAX2 easily traverses firewalls and network address translators. This is in contrast to SIP, H.323 and MGCP which use an out-of-band RTP stream to deliver information.

IAX2 supports trunking, multiplexing channels over a single link. When trunking, data from multiple calls are merged into a single stream of packets between two endpoints, reducing the IP overhead without creating additional latency. This is advantageous in VoIP transmissions, in which IP headers use a large percentage of bandwidth.
[edit] Origin of IAX

The IAX and IAX2 protocols were created by Mark Spencer for Asterisk for VoIP signaling. The protocol sets up internal sessions and these sessions can use whichever codec they want for voice transmission. The Inter-Asterisk Exchange protocol essentially provides control and transmission of streaming media over IP (Internet Protocol) networks. IAX is extremely flexible and can be used with any type of streaming media including video, however it is mainly designed for control of IP voice calls.
[edit] The goals of IAX

The primary goals for IAX were to minimize bandwidth used in media transmissions, with particular attention drawn to control and individual voice calls, and to provide native support for NAT (Network Address Translation) transparency. Another goal is to be easy to use behind firewalls.

The basic structure of IAX is that it multiplexes signaling and multiple media streams over a single UDP (user datagram protocol) stream between two computers. IAX is a binary protocol, designed to reduce overhead especially in regard to voice streams. Bandwidth efficiency in some places is sacrificed in exchange for bandwidth efficiency for individual voice calls. One UDP stream is easier to setup for users that are behind a firewall.

An additional benefit to having a single stream is the added security, which can be implemented very easily. Furthermore, in countries where ISPs are filtering VoIP, IAX can be easily hidden.
[edit] IAX drawbacks

* Awkward extensibility: Due to the lack of a generic extension mechanism, every new feature has to be added in the protocol specification, which makes it less flexible than H.323, SIP or MGCP.
* Vulnerability: IAX2 is vulnerable to Resource Exhaustion DoS 0days that are currently available to the public. There are currently no solutions to these issues. The current best practices include limiting UDP port access to specific trusted IP addresses. Internet facing IAX2 ports are considered vulnerable and should be monitored closely. The fuzzer used to detect these application vulnerabilities was posted on milw0rm and is included in the VoIPer svn tree. These issues were briefly mentioned in the IAX RFC #5456 on page 94.