Section 7.8. Key Issues: Replacing Call Signaling with VoIP

7.8. Key Issues: Replacing Call Signaling with VoIP

  • VoIP can replace phone-to-phone signaling that the PBX provides in traditional environments

  • Legacy protocol support is still required for analog phones and connections to the PSTN

  • The two most popular standards for phone-to-phone signaling are H.323 and SIP

  • H.323 was created by the ITU as a video conferencing standard but grew to become an ambitious PBX-replacement recommendation

  • Microsoft NetMeeting and OhPhone are H.323 softphones

  • A gatekeeper and gateway form the softPBX nucleus on an H.323 network. These two elements often run on a single server

  • H.245, H.225, and RAS are the three layers of a phone-to-phone signaling session on an H.323 network

  • SIP was created by the IETF as a media-session management protocol; it has proven a great match for telephony applications on the Internet

  • SIP defines less of the network than H.323 does, leaving to the application developer the details of application, session, and presentation layers

  • SIP doesn't address legacy interfacing at all

  • SIP is seen by many PBX vendors as a way of signaling trunk connections to other vendors ' equipment

  • SDP is SIP's capabilities negotiation protocol

  • RTP is the packetization and framing mechanism used by SIP and H.323

  • IAX, MEGACO/H.248, and Cisco SCCP are other prevalent signaling protocols

  • IAX does not use RTP for packetization; it frames signaling and sound data in the same packet construct

  • Heterogeneous signaling is required when an endpoint of one signaling protocol wishes to communicate with an endpoint of another

Switching to VoIP
Switching to VoIP
ISBN: 0596008686
EAN: 2147483647
Year: 2005
Pages: 172

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