VigorBX 2000 Series User’s Guide
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IP PBX integrates the benefits of VoIP and transfers the message from IP phone into the data
that can be accepted by traditional PBX through IP network. It is a new platform that
enterprises can use data network to deliver voice. Additionally, to move the IP phone set(s),
users just need to plug into another network connector. Such thing simplifies the procedure
of moving, increasing, changing and deleting phone settings; also it can join with other
system such as CALL center to be a multi-functional communication platform. Moreover, it
can save large cost in communication for the enterprise.
Voice over IP network (VoIP) enables you to use your broadband Internet connection to make
toll quality voice calls over the Internet.
Info
This function is used for “V” models.
There are many different call signaling protocols, methods by which VoIP devices can talk to
each other. The most popular protocols are SIP, MGCP, Megaco and H.323. These protocols
are not all compatible with each other (except via a soft-switch server).
The Vigor V models support the SIP protocol as this is an ideal and convenient deployment for
the ITSP (Internet Telephony Service Provider) and softphone and is widely supported. SIP is
an end-to-end, signaling protocol that establishes user presence and mobility in VoIP
structure. Every one who wants to talk using his/her SIP Uniform Resource Identifier, “SIP
Address”. The standard format of SIP URI is
sip: user:password @ host: port
Some fields may be optional in different use. In general, "host” refers to a domain. The
“userinfo” includes the user field, the password field and the @ sign following them. This is
very similar to a URL so some may call it “SIP URL”. SIP supports peer-to-peer direct calling
and also calling via a SIP proxy server (a role similar to the gatekeeper in H.323 networks),
while the MGCP protocol uses client-server architecture, the calling scenario being very
similar to the current PSTN network.
After a call is setup, the voice streams transmit via RTP (Real-Time Transport Protocol).
Different codecs (methods to compress and encode the voice) can be embedded into RTP
packets. Vigor V models provide various codecs, including G.711 A/
μ
-law, G.723, G.726 and
G.729 A & B. Each codec uses a different bandwidth and hence provides different levels of
voice quality. The more bandwidth a codec uses the better the voice quality, however the
codec used must be appropriate for your Internet bandwidth.
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