Draytek 2800 Series User Guide

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Vigor2800 Series User’s Guide 
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Voice over IP network (VoIP) enables you to use your broadband Internet connection to 
make toll quality voice calls over the Internet.   
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.     
Usually there will be two types of calling scenario, as illustrated below: 
Calling via SIP Servers   
 
First, the Vigor V models of yours will have to register to a SIP Registrar by sending 
registration messages to validate. Then, both parties’ SIP proxies will forward the 
sequence of messages to caller to establish the session. 
 
If you both register to the same SIP Registrar, then it will be illustrated as below: 
 
 
 
The major benefit of this mode is that you don’t have to memorize your friend’s IP 
address, which might change very frequently if it’s dynamic. Instead of that, you will