IP Telephony/VoIP Protocols - NetwaxLab

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Monday, July 13, 2015

IP Telephony/VoIP Protocols

Standardized
Proprietary
H.323
Skype
SIP
Cisco System (Skinny)
H.248
Asterisk (IAX, open source), etc
 
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


Audio/Video Transport
Signaling
ITU-T
RTP/RTCP
H.323
H.248

IETF

RTP/RTCP
SIP
MGCP
MEGACO

Standard protocols: Media Transport

  • Protocols designed to deliver real time data to the remote entities:
    • 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
    • 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)
RTCP
RTP
UDP
Network Protocol and Network Interface

Standard Protocols: Call Signaling

  • Before audio or video media can flow using RTP/RTCP between two entities
    • 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
    • H.323 (ITU-T Study Group 16, version 5, 2003)
    • 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
    • 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)

Standard Protocols: Additional Signaling

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

Specifications

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