Understanding SIP Phones in Telephony

“Soft” Telephony

SIP Phones • software-based: an application which makes use of your computer’s microphone and speakers or an attached headset to allow you to make or receive calls • hardware-based: looks and behaves just like a normal phone but is connected directly to the data network, rather than to standard PSTN line(s)

… • an ATA adapter allows you to plug the Ethernet network jack into the adapter and then plug the phone into the adapter; the old phone will appear to the VoIP phone system software as a regular SIP phone • a broadband connection and connection to a VoIP provider or a SIP Server are required • software SIP Phones can also run on mobile devices such as Android and iOS

Commonality • An Internet telephony service provider (ITSP), also known as voice service providers (VSP), offers digital telecommunications services based on Voice over Internet Protocol (VoIP) that are provisioned via the Internet • Both end-points must support the same Voice-over-IP protocol • Both must have at least one common audio codec

Protocols • Session Initiation Protocol (SIP): standardized by the Internet Engineering Task Force (IETF) • Skype: uses proprietary protocols • Google Talk: uses the Extensible Messaging and Presence Protocol (XMPP) • Inter-Asterisk eXchange protocol (IAX): from the open-source software application Asterisk • H.323: one of the earliest VoIP protocols; use is declining, rarely used for consumer products

Session Initiation Protocol (SIP) • a communications protocol for signaling, for the purpose of controlling multimedia communication sessions • Internet telephony, business IP telephone systems, service providers and all of the carriers use SIP • SIP can be used to set up and control voice and video calls, as well as instant messaging

… • most common application of SIP: the setup and termination of Voice over IP (VoIP) telephone calls • inform the calling party of the Internet Protocol (IP) address of the called party's telephone • data units containing segments of digitized speech may be then transmitted to the called party's telephone, implementing Voice over IP (VoIP) speech communication

… • other protocols specify the media format and protocol to be used to communicate the media • carry a Session Description Protocol (SDP) message specifying the codec and the use of either the Real-time Transport Protocol (RTP) or Secure Real-time Transport Protocol (SRTP) • RTP Protocol data units may be encrypted byTransport Layer Security (TLS) for secure transmission

History • originally designed by Mark Handley, Henning Schulzrinne, Eve Schooler and Jonathan Rosenberg in 1996 • standardized as RFC 2543 in 1999 • revised in June 2002 in RFC 3261 • extended for video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer, fax over IP and online games

… • distinguished for having roots in the Internet community rather than in the telecommunications industry • standardized primarily by the Internet Engineering Task Force (IETF), versus other International Telecommunication Union (ITU) protocols such as H.323

Focus • 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) • by itself does not define these features: familiar telephone-like operations (i.e. dialing a number, causing a phone to ring, hearing ringback tones or a busy signal) are performed by proxy servers and user agents

… • Implementation and terminology are different in the SIP world compared to the PSTN but, to the end-user, the behavior is similar • only involved in the signaling portion of a media communication session, primarily used to set up and terminate voice or video calls • can be used to establish two-party (unicast) or multiparty (multicast) sessions

… • allows modification of existing calls: changing addresses or ports, inviting more participants, and adding or deleting media streams • For call setup, the body of a SIP message contains a Session Description Protocol (SDP) data unit, which specifies the media format, codec and media communication protocol • Voice and video mediatypically use Real-time Transport Protocol (RTP) or Secure Real-Time Transport Protocol (SRTP)

Network Elements • A user agent is a logical network end-point used to create or receive SIP messages; as a client (UAC), it sends SIP requests, and as a server (UAS) it receives requests and returns a SIP response • A SIP phone is an IP phone that implements client and server functions of a SIP user agent and provides the traditional call functions of a telephone, such as dial, answer, reject, call hold, and call transfer

… • A proxy server is a network server with UAC and UAS components that functions as an intermediary entity for the purpose of performing requests on behalf of other network elements. A proxy server primarily plays the role of routing, meaning that 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, such as for determining whether a user is allowed to make a call

… • A registrar is a SIP endpoint provides a location service. It accepts REGISTER requests, recording the address and other parameters from the user agent. For subsequent requests it provides an essential means to locate possible communication peers on the network

… • A redirect server is a user agent server that generates 3xx (redirection) responses to requests it receives, directing the client to contact an alternate set of URIs. A redirect server allows proxy servers to direct SIP session invitations to external domains

… • Session border controllers serve as middle boxes between UA and SIP servers for various types of functions, including network topology hiding and assistance in NAT traversal • Gateways can be used to interconnect a SIP network to other networks, such as the public switched telephone network, which use different protocols or technologies

Media Gateway Control Protocol (MGCP) • A text-based protocol consisting of commands and responses. It uses the Session Description Protocol (SDP) for specifying and negotiating the media streams to be transmitted in a call session and the Real-time Transport Protocol (RTP) for framing the media streams • a master/slave protocol that allows a call control device such as a Call Agent to take control of a specific port on a media gateway

… • Reflects the structure of the PSTN, with the power of the network residing in a call control center softswitch, which is analogous to the central office in the telephone network • The endpoints are low-intelligence devices, mostly executing control commands and providing result indications in response • Gateway Control Protocol (H.248, Megaco) implementation

Understanding SIP Phones in Telephony