Quintum Technologies Tenor Call Relay 60 - Manual (Page 13)

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Chapter 1: Overview
Remove Unnecessary Call Delay
Since redundant decompression and re-compressions processes are eliminated as a result of Call
Relay 60 (linking two VoIP networks via PSTN gateways is not required); this improves voice qual-
ity by removing unnecessary delay, latency, and distortion.
Figure 1-5
IP Network
Call Relay Call Relay
VoIP
Endpoint
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VoIP
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VoIP Network A VoIP Network B
IP Used to Link
Networks
Eliminates
redundant encoding
and decoding
IP Network IP Network
Call Relay Call Relay
VoIP
Endpoint
VoIP
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VoIP
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VoIP
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VoIP
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VoIP
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VoIP
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VoIP
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VoIP
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VoIP
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VoIP
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VoIP
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VoIP
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VoIP
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VoIP
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VoIP
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VoIP
Endpoint
VoIP
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VoIP
Endpoint
VoIP
Endpoint
VoIP
Endpoint
VoIP
Endpoint
VoIP
Endpoint
VoIP
Endpoint
VoIP Network A VoIP Network B
IP Used to Link
Networks
Eliminates
redundant encoding
and decoding
Call Relay 60 Call Relay 60
Removes Unnecessary Call Delay
Voice Control Management
H.323 Gatekeeper
The Call Relay 60 complies with the H.323 industry specifications for voice control and manage-
ment. It performs IP call routing functions (for calls entering and exiting a site). The Gatekeeper
(internal to a Call Relay 60) collects, manages, and distributes call routing information. The Border
Element (internal to the Call Relay 60) provides access into or out of an administrative domain.
Other H.323 endpoints, such as gateways, can register to the internal Gatekeeper.
Session Initiation Protocol (SIP)
SIP (Session Initiation Protocol) is a signaling protocol used to establish a session on an IP network
for voice control and management; it is a request-response protocol that closely resembles Hypertext
Transfer Protocol (HTTP), which forms the basis of the World Wide Web. SIP re-uses many of the
constructs and concepts of Internet protocols such as HTTP and Simple Mail Transfer Protocol
(SMTP). The purpose of SIP is only to establish/change/terminate sessions. SIP is not concerned
with the content or details of the session.
SIP is Transport layer-independent, which means it can be used with any transport protocol: UDP,
TCP, ATM, etc. It is text-based, so it requires no encoding/decoding like H.323. And SIP supports
user mobility, using proxies and redirecting requests to your current location.
When configured for SIP the Call Relay 60 will act as a SIP User Agent (Endpoint) as defined in
IETF RFC3261. Multiple user agents allow for separate agents to be allocated to each SIP call. It
will be able to direct calls to and from the IP network, and Customer Premise Equipment (CPE) such