Voice over Internet Protocol (VoIP) requires several specialized protocols to deliver voice communications between...
By submitting your personal information, you agree that TechTarget and its partners may contact you regarding relevant content, products and special offers.
all parties on a call. But because VoIP delivers those communications over a network or the Internet, it also relies indirectly on network protocols to facilitate operations.
In this VoIP protocol listing, we look at more than 30 protocols that make VoIP run.
Use the table of contents below to navigate to the VoIP protocol of interest or peruse the listing below that.
Common Open Policy Service Protocol (COPS) is a proposed standard protocol for distributing network policy information from a policy decision point, such as a network server, to policy enforcement points. COPS is used to assign priorities for voice, video, and other high-bandwidth traffic as part of Quality of Service (QoS) for VoIP connections.
Communications over IP (CoIP) is a set of standards that define the transmission of voice, video, textual and any other type of digital communication over the Internet. CoIP is considered an expansion of VoIP. The abbreviation CoIP also refers to Conferencing over IP.
Differentiated Services (DiffServ or DS) is a method for providing different levels of service to network traffic based on Class of Service or QoS settings. Voice traffic, for example, requires a certain amount of bandwidth, low delay and low jitter, so it would receive precedence over other types of traffic if DiffServ is implemented. DiffServ manages bandwidth allocation for VoIP connections as well as other Internet media connections.
Distributed Universal Number Discovery (DUNDi) is a peer-to-peer system for translating phone numbers and internal extensions to VoIP addresses. Each DUNDi client has to be aware of at least one other DUNDi client, and one of the peers must know how to reach the desired extension or number. DUNDi provides directory services that may be accessed by VoIP protocols such as Session Initiation Protocol (SIP) and H.323 to help route calls from their initiators to their proper recipients.
Dynamic Host Configuration Protocol (DHCP) is a communications protocol that supports assignment of IP addresses to network clients automatically. This enables an administrator to move a computer from one location to another, and the computer automatically receives a new IP address to connect to the network. The alternative to DHCP is the manual assignment of static IP addresses. DHCP leases IP addresses to clients; the leases have time limits and can be renewed. DHCP can be used to assign IP addresses to VoIP phones.
Extensible Messaging and Presence Protocol (XMPP) is a protocol based on Extensible Markup Language that is used for exchanging near-real-time communications, such as instant messages and other request-response services, and for detecting online presence. As XMPP evolved, it added an extension for media session services (called Jingle), which provides VoIP calling on top of XMPP.
H.245 is a protocol used to control the setup and closure of media channels in packet-based networks for audio, video, voice and data transmissions. It also transmits information needed for flow control, jitter management and encryption. H.245 is used within H.323 sessions.
H.248, also known as Megaco or Media Gateway Control Protocol (MGCP), is a protocol used to handle both signaling and session management for multimedia conferencing. H.248 enables a media gateway to manage communications between circuit-switched networks and packet-switched networks, and may also be used to set up, manage and terminate calls among multiple endpoints. Originally defined in RFC 2705, MGCP is a master-slave protocol that allows ports on a media gateway to be controlled by a call agent (in this case, the media gateway controller). MGCP lets the controller determine the location and media-handling capabilities for all endpoints in a conference to select a level of service that works for all participants. Megaco/H.248 refines MGCP to support more ports (participants) per gateway, and adds TDM and ATM communication modes to conferencing capabilities.
H.323 is a packet-based standard that supports audio, video and data communications across IP networks. It supports point-to-point and multipoint conferencing, as well as the gateway administration of media traffic, bandwidth and user participation. H.323 provides several functions required by VoIP, including call control, multimedia management and bandwidth management. H.323 is the most widely used VoIP protocol in the enterprise.
Integrated Services Digital Network (ISDN) is a set of standards that supports digital telephone service and other data transmissions over existing copper telephone wiring. At 128 Kbps, the first ISDN Internet connections were double the speed of ordinary dialup modems. Some medium-sized and large organizations use ISDN30 (a version of ISDN) with a private branch exchange (PBX) telephone system to deliver voice and data. ISDN30 service has a bandwidth of 2 Mbps. VoIP can be run over ISDN to achieve low latency and to provide necessary QoS for VoIP.
Inter-Asterisk eXchange (IAX) Protocol (pronounced "eeks") is a communications protocol used to create interactive user sessions on IP networks, allowing data to be transmitted through several channels over a single link. IAX is used for controlling the transmission of streaming media and VoIP. IAX is an open source initiative developed as an alternative to several VoIP-related protocols, such as SIP, MGCP and Real-time Transport Protocol (RTP). Compared with SIP, for example, IAX is more efficient at using bandwidth, but isn't as popular. SIP was standardized long ago and is in wide use. IAX suffers from some limitations, such as the use of a single port and the lack of a generic extension mechanism.
Internet Protocol (IP) is one of the main protocols used to send data over the Internet and private networks. Each host on a network is assigned an IP address, which uniquely identifies it from all other hosts. VoIP transmits voice data (phone calls) over IP networks.
Jingle protocol is a signaling protocol that enables XMPP entities to set up, manage and tear down multimedia sessions. More specifically, Jingle is an extension of XMPP and provides VoIP calling on top of XMPP.
Media Gateway Control Protocol (MGCP) See H.248.
Message Session Relay Protocol (MSRP) is a protocol for exchanging instant messages or chat data between two or more parties across an IP network. Messages exchanged between peers are referred to as an MSRP session. MSRP works with SIP signaling, in that a session can start with a SIP INVITE transaction, which carries a Session Description Protocol (SDP) offer/answer exchange to establish the session. The MSRP session can be terminated with a SIP BYE request.
For more information on this VoIP protocol listing
View Chapter 3 of the book Beyond VoIP Protocols, republished with permission by John Wiley & Sons Inc.
Learn the security weaknesses of VoIP protocols.
Security expert Lisa Phifer explains VoIP protocol insecurities in this tip.
Open Settlement Protocol (OSP) is a client-server protocol that manages access control, accounting, usage data and interdomain routing for Internet service providers to support IP telephony. Through OSP, VoIP service providers can use services from OSP clearinghouses, support secure VoIP peering and arrange end-to-end telephony services across multiple voice and/or data networks.
Q signaling (QSIG) is an Integrated Services Digital Network (ISDN) protocol used for interconnecting digital PBXs. QSIG is based on the Q.931 standard and ensures that call setup, call processing and other essential functions are carried out.
Q.931 is a standard signaling protocol for the ISDN communications used in VoIP. The Q.931 protocol is involved in setting up, transmitting and receiving call signaling messages and terminating call connections. Q.931 is part of the VoIP H.323 protocol stack, used to manage ISDN connection control signaling.
Real-Time Transport Protocol (RTP) is an Internet protocol used to send and receive multimedia information between any two devices; that is, RTP defines how to deliver audio and video over IP networks. VoIP uses RTP and SIP to package and stream multimedia content.
Real-Time Transport Control Protocol (RTCP) is a protocol that works with the RTP to monitor data delivery on large multicast networks, mainly for streaming media, telephony and video conferencing. RTCP sends control packets to participants in a streaming multimedia session, whereas RTP delivers the data. The purpose of monitoring is to determine whether RTP is providing the necessary QoS and to compensate for delays, if needed.
Resource Reservation Protocol (RSVP) is a protocol that sets aside resources, such as Internet paths, for the transmission of high-bandwidth data. For example, a client can send an RSVP request to receive a video (multicast) before the video begins. The reservation for sufficient bandwidth and priority packet scheduling is made through all gateways, all the way to the destination, providing a good user experience while the video plays. RSVP is used on VoIP networks.
Secure Real-Time Transport Protocol (SRTP) is an extension to RTP and adds encryption and authentication for enhanced security, especially against denial-of-service attacks. SRTP was designed primarily for VoIP communications.
Session Announcement Protocol (SAP) is a protocol used to distribute information about multicast conferencing sessions. To announce a multicast session, the session creator sends a multicast packet to a well-known multicast group. Participants listen to the group and receive its multicast packets. SAP is used to define the format for the multicast packets and works with the SDP to describe packets. SAP and SDP are used by VoIP signaling protocols such as SIP and H.323.
Session Description Protocol (SDP) is a protocol used to distribute information about multicast conferencing sessions. SDP defines a format used to describe multimedia sessions. VoIP signaling protocols such as SIP and H.323 use SDP.
Session Initiation Protocol (SIP) is a signaling protocol used to set up VoIP connections. SIP was designed as an alternative to H.323 signaling. The most common use of SIP is for setting up and tearing down voice and video calls, but it can also be used to apply modifications to existing calls, such as adding or removing call participants. SIP uses a request-response model, so each SIP transaction consists of a client request, followed by at least one server response.
Signaling System 7 (SS7) is a standard that defines how network elements in a public switched telephone network (PSTN) exchange information over a digital signaling network for greater efficiency and security. More specifically, SS7 describes how to place telephone call setup and management information in a channel/network other than the originating channel/network, referred to as out-of-band signaling. Moving calls between SS7 and VoIP environments requires a gateway to mediate between those two environments.
Skinny Client Control Protocol (SCCP) is a Cisco proprietary standard for allowing "skinny" clients to communicate with H.323 VoIP systems. The term skinny refers to a lightweight protocol that requires minimal computer processing. SCCP places the required H.323 call setup processing capabilities in a proxy called a call manager. The skinny client and the call manager use SCCP to communicate with each other over User Datagram Protocol (UDP) and IP.
Skype protocol is a peer-to-peer Internet telephony protocol created specifically for Skype sessions; it does not work with most standard VoIP networks without licensing from Skype. Skype requires that Ports 80 and 443 be open for outgoing Transmission Control Protocol (TCP) transmissions, and recommends that all destination ports above 1024 also remain open. In addition, Ports 5060 and 8000 should remain open for incoming and outgoing UDP transmissions. Skype prefers UDP for voice transmissions.
T.38 is a protocol for sending faxes over a VoIP network or the Internet in real time. T.38 most often uses UDP but can use TCP as well. UDP is preferred to avoid delays, since UDP sends redundant data packets to deal with possible packet loss.
Transmission Control Protocol/Internet Protocol (TCP/IP) is the name of the standard protocols and services in use on the Internet and many internal networks. TCP is a robust, reliable, connection-oriented protocol that manages the disassembly of messages into smaller packets that are then transmitted. A TCP layer on the receiving end of the transmission reassembles the packets into the original message. IP is the protocol that handles the routing and delivery of packets. VoIP voice and signaling communications are sent using standard TCP/IP protocols.
User Datagram Protocol (UDP) is a communications protocol used to deliver messages over a network or the Internet that doesn't require sequencing or error checking. UDP is an alternative to TCP. With UDP, messages are not disassembled and reassembled, so the application delivering the message is responsible for ensuring the entire message has arrived intact. VoIP often uses UDP to send voice transmissions.
Voice over Internet Protocol (VoIP) is a collection of related technologies, protocols and techniques used to deliver voice communications over an IP network, such as the Internet. With VoIP, voice information is sent in digital packets over an IP network, like the Internet, rather than over the PSTN. One of the primary benefits of VoIP is reduced costs for telephone service.
For more information, view our webpage on VoIP protocols and standards.
About the VoIP protocol listing authors
Ed Tittel is a full-time freelance writer, trainer and consultant who's written more than 100 books, including his latest, The PC Magazine Guide to Fighting Spyware, Viruses and Malware. Ed has been active in the computing industry for more than 20 years as a software developer, manager, writer and trainer.
Kim Lindros has more than 15 years of experience in the computer industry, from technical support specialist to network administrator to book and course content manager. She has edited and developed more than 150 IT-related books and online courses, and co-authored two certification books and numerous online articles with Ed. Kim runs Gracie Editorial, a content development company.