Cisco Systems H.323/SIP Manual De Usuario
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Administrator Guide for Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1
OL-6571-02
Chapter 1 Introducing Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1
Information About Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Components
Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1
IP telephony uses your data network infrastructure to transmit voice packets. The underlying technology
that is used by IP telephony applications is Voice over IP (VoIP), which enables different types of
endpoints—IP phones, PSTN phones, and H.323 clients, for example—to communicate over your
network.
that is used by IP telephony applications is Voice over IP (VoIP), which enables different types of
endpoints—IP phones, PSTN phones, and H.323 clients, for example—to communicate over your
network.
The following sections provide information about VoIP concepts and how they relate to Release 5.2.1:
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Protocols That Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Uses,
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Standards That are Supported by Cisco Unified MeetingPlace H.323/SIP IP Gateway Software
Release 5.2.1
Release 5.2.1
Release 5.2.1 supports the following networking and telephony standards:
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H.323
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SIP
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RTP
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Codec G.711 alaw and ulaw (64 kbps) and G.729a (8 kbps)
Note
By default, G.729a is not enabled, and G711 codec calls are negotiated first. For more
information about assigning codec preferences, see the Configuration Guide for Cisco Unified
MeetingPlace Audio Server Release 5.3 at the following URL:
information about assigning codec preferences, see the Configuration Guide for Cisco Unified
MeetingPlace Audio Server Release 5.3 at the following URL: