IP Telephony/VoIP Protocols

Standardized
H.323
SIP
H.248

Proprietary
Skype
Cisco System (Skinny)
Asterisk (IAX, open source), etc

Figure 1 VoIP

Protocol Analyzer

Proprietary Protocols

have restricted innovation (smaller users/developers community, narrowed vision, solving

smaller issues)
restrict the set of available functionalities because of interoperability (developing gateways to
every protocol take too long)

Standardized Protocols
ITU-T

Audio/Video Transport
RTP/RTCP

IETF

RTP/RTCP

Signaling
H.323
H.248
SIP
MGCP
MEGACO

IP Telephony/VoIP Protocols
Standard protocols: Media Transport

Protocols designed to deliver real time data to the remote entities:

o RTP (Real Time Protocol: IETF RFC 3550, July 2003)
Provides end-to-end network transport functions suitable for applications
transmitting real time data, such as audio, video
o RTCP (Real Time Control Protocol: IETF RFC 3550, July 2003)
Control protocol to allow monitoring of the data delivery, and to provide minimal
control and identification functionalities
RTP/RTCP are always sent on top of UDP (User Datagram Protocol) on IP-based networks
Codec (G.7xx, GSM, iLBC, Speex, H.26x)
RTP

RTCP

UDP
Network Protocol and Network Interface

Standard protocols: Call Signaling

Before audio or video media can flow using RTP/RTCP between two entities
o need of finding the remote device and to negotiate the means by which media will flow
between the two devices
The protocols that are central to this process are referred to as call signaling protocols, the two
standardized are
o H.323 (ITU-T Study Group 16, version 5, 2003)
o SIP (Session Initiation Protocol, original IETF RFC 2543, updated by IETF RFC 3261, June
2002)

A little bit of story:

H.323 and SIP both have their origins in 1995

H.323 enjoyed the first commercial success due to the fact that ITU quickly published the first
standard in early 1996
SIP progressed much more slowly in the IETF with the first recognized "standard" published later
in 1999
SIP was revised over the years and re-published in 2002 as RFC 3261, which is the currently
recognized standard for SIP
These delays in the standards process resulted in delays in market adoption of the SIP protocol
Today H.323 is still having the bigger commercial market share but the trend is toward SIP
o SIP was chosen as the official protocol by the 3GPP partnership alliance for the UMTS IMS
(IP Multimedia subsystem) (SIP is the official protocol for IP-based call signaling in UMTS
first application: Push To Talk (PTT)

IP Telephony/VoIP Protocols
Standard Protocols: Additional Signaling

MGCP (Media Gateway Control Protocol: IETF RFC 3661, was RFC 3435, was RFC 2705)
o control protocol for controlling Media Gateways (MG) from external call control elements
called Media Gateway Controllers (MGC)
MEGACO (MEdia GAteway COntrol protocol)
o This version of the protocol is the next generation of MGCP
o Joint effort of the IETF MEGACO working group and the ITU Study Group 16
IETF refer to the protocol as MEGACO (RFC 3525, was RFC 3015, was RFC 2885)
ITU refers to it as H.248
o Currently is under discussion the MEGACO/H.248 version 2
Summarizing: they are not IP Telephony protocols of their own!
o they are addressing complementary topics related to media control on gateways (only
legacy voice features)
o need to use them to achieve IP Telephony

Overview of Protocol
H.323
H.323 was the first of the four voice signaling protocols and definitely has maturity on its side. The
International Telecommunication Union, Telecommunication Standardization Sector (ITU-T) originally
created H.323 to allow simultaneous voice, video, and data to transmit across ISDN connections. It has
since been adapted to work over LAN environments.
Session Initiation Protocol (SIP)
SIP is often called the next generation of H.323. Developed by the Internet Engineering Task Force
(IETF), SIP is a much more lightweight and scalable protocol than H.323. While support for SIP is
widespread, it is an evolving standard that does not currently support many of the advanced features
232 CCNA Voice Official Exam Certification Guide of VoIP networks. As SIP becomes more mature and
robust, it is poised to become the primary VoIP signaling standard used worldwide (similar to the way
data networks use TCP/IP today).
Media Gateway Control Protocol (MGCP)
MGCP is the first true client/server VoIP signaling protocol. If you are using MGCP, you will perform
the vast majority of your gateway configuration from a centralized system known as a call agent.
Because this is a newer IETF standard, it is not as widely supported as H.323 or SIP.
Skinny Client Control Protocol (SCCP)
SCCP is the only Cisco-proprietary VoIP protocol currently in use. Although SCCP is not specifically
designed for gateway signaling and control, a limited number of Cisco gateways do support it. The

IP Telephony/VoIP Protocols
primary goal of SCCP is to provide a signaling protocol between the Cisco Unified Communications
Manager and Cisco IP phones. Similar to MGCP, the SCCP devices report every action to the
Communications Manager server, which then responds with the action the device should take.
RTP
The Real-time Transport Protocol (RTP) provides end-to-end network transport functions suitable for
applications transmitting real-time data such as audio, video or simulation data, over multicast or
unicast network services. RTP does not address resource reservation and does not guarantee quality-ofservice for real-time services. The data transport is augmented by a control protocol (RTCP) to allow
monitoring of the data delivery in a manner scalable to large multicast networks, and to provide minimal
control and identification functionality. RTP and RTCP are designed to be independent of the underlying
transport and network layers. The protocol supports the use of RTP-level translators and mixers.
RTCP
The RTP Control Protocol (RTCP) is based on the periodic transmission of control packets to all
participants in the session, using the same distribution mechanism as the data packets. The underlying
protocol must provide multiplexing of the data and control packets, for example using separate port
numbers with UDP.

Specification

Figure 2 Protocol

Specification