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608 Chapter 16: Voice Techniques
Voice over IP
Of the emerging technologies, voice over IP is the most sensitive. Voice design sets the standard
for engineering and quality in a network. Demand for voice over IP is leading the movement
for QoS in IP environments. It will ultimately lead to the use of the Internet for fax and voice
telephony services. Voice over IP will ultimately be a key component of the migration of
telephony to the LAN and WAN infrastructure.
VoIP Signaling
VoIP signaling has three major areas:
·
Signaling from the PBX to the router
·
Signaling between routers
·
Signaling from the router to the PBX
The user's phone is connected to the PBX. When the user picks up the handset, an off-hook
condition is generated. The off-hook condition generates a signal from the PBX to the router.
The connection between the PBX and the router appears as a trunk line to the PBX, which
signals the routers to seize the trunk. Signaling from the PBX may be any of the common
signaling methods used to seize a trunk line, such as FXS or E&M signaling. In the future,
digital signaling such as CCS or QSIG will become available. The PBX then forwards the
dialed digits to the router in the same manner the digits would be forwarded to a telephone
company switch. Within the router, the dial plan mapper maps the dialed digits to an IP address
and signals a Q.931 Call Establishment Request to the remote peer that is indicated by the IP
address. Meanwhile, the control channel is used to set up the Real-Time Protocol (RTP) audio
streams, and the RSVP protocol is used to request a guaranteed quality of service.
When the remote router receives the Q.931 call request, it signals a line seizure to the PBX.
After the PBX acknowledges, the router forwards the dialed digits to the PBX and signals a call
acknowledgment to the originating router.
In connectionless network architectures such as IP, the responsibility for session establishment
and signaling reside in the end stations. To successfully emulate voice services across an IP
network, enhancements to the signaling stacks are required. For example, an H.323 agent is
added to the router for standards-based support of the audio and signaling streams. The Q.931
protocol is used for call establishment and teardown between H.323 agents or end stations. The
Real-Time Control Protocol (RTCP) is used to establish the audio channels themselves. A
reliable session-oriented protocol, TCP, is deployed between end stations to carry the signaling
channels. RTP, which is built on top of UDP, is used to transport the real-time audio stream.
RTP uses UDP as a transport mechanism because it has lower delay than TCP and because
actual voice traffic, unlike data traffic or signaling, tolerates low levels of loss and cannot
effectively exploit retransmission.
87200333.book Page 608 Wednesday, August 22, 2001 1:41 PM