ch03s04.html

IP_Telephony_Cookbook主要讲解的是IP电话方面的知识,对这个方面需求的读者会很有帮助

alt="Integrated Voice and Video over IP architecture at SURFnet offices" 
src="ch03s04.files/vvoip.png" align=middle></DIV></DIV>
<P>The goal of the integration of voice and videoconferencing over IP was to 
make possible to refer directly to the user without knowing his location and 
what terminal is actually using. When someone contacts the user by any means 
(PSTN or H.323 of H.320), the call should be completed by reaching any device 
the user may have operational. The components necessary to establish such an 
integrated infrastructure are:</P>
<DIV class=itemizedlist>
<UL type=disc>
  <LI>a PBX, connected to the PSTN, in this case a Philips Sopho, to handle all 
  incoming regular voice calls;
  <LI>a H.323 gateway, in this case a Radvision L2W (LAN to WAN H.323) gateway, 
  on the one side connected to the PBX (by 2 ISDN lines) and on the other side 
  to the local IP network;
  <LI>a H.323 gatekeeper, in this case the build-in gatekeeper of the Radvision 
  gateway, to route all calls on the IP network, including making decisions when 
  to route the call off-net (to the PSTN through the PBX);
  <LI>a Callmanager, in this case a Cisco Callmanager, being the gatekeeper to 
  performing PBX-like functions for the IP-phones;
  <LI>endstations, being the user's terminal(s). </LI></UL></DIV>
<P>The terminals used here are:</P>
<DIV class=itemizedlist>
<UL type=disc>
  <LI>IP phones, in this case Selsius and Cisco IP phones, registered on the 
  Callmanager;
  <LI>H.323 videoconferencing stations, in this case VCON EscortPro boards in 
  PCs with Meetingpoint 4.6 and Polyspan Viewstations (128 and FX), registered 
  at the H.323 gatekeeper;
  <LI>regular DECT phones, in this case Philips, registered at the 
PBX;</LI></UL></DIV>
<P>For allowing multipoint calls a MCU (Radvision MCU323) has to be added and 
the conference feature on the PBX to be enabled. These are not necessary 
functionalities, but can enhance the communication experience.</P>
<P>The means by which integration was established was the Dial Plan that 
guaranteed unique number addressing for all devices. The Global Dialing System 
(GDS, see section on <A title="7.2.2.1.&nbsp;Global Dialing Scheme" 
href="http://www.informatik.uni-bremen.de/~prelle/terena/cookbook/main/ch07s02.html#gds">dialplans</A> 
and <A href="http://www.wvn.ac.uk/support/h323address.htm" 
target=_top>http://www.wvn.ac.uk/support/h323address.htm</A>) was adopted, and 
the full E.164 addressing, number of videoconferencing and IP telephony endpoint 
numbering allow all terminals to be used like regular phones. Thereby it is 
guaranteed that for terminals called/dialed from the PSTN, the call would be 
routed to the PBX. Also the other way around, from the voice and video over IP 
terminals any regular PSTN number could be dialed, without the need for 
rewriting the dial string. GDS is supported by the ViDeNet H.323 gatekeeper 
hierarchy, which resembles the phone system in that it is a hierarchy of 
distributed gatekeepers that route IP calls based on prefix resolution.</P>
<P>In the examples below, "A"'s phone number is 030-2305367, and his 
international is 0031302305367. His IP phone number is 030-2305167. For 
demonstration purposes he has also registered his H.323 station as 030-2305367. 
00 is the international access code in the Netherlands, 030 is the area code of 
Utrecht, 23053.. and 23051.. are the prefixes/numberblocks SURFnet has control 
of and 67 is the local office number assigned to "A" (Note: to "A" and not his 
devices, because there is more than 1 numberblock that holds 67 (367 and 
167).</P>
<P>"A" registers his H.323 station with the number 67 at SURFnet's office 
gatekeeper, which itself is registered with prefixes 3023051 and 3023053 at the 
Dutch national gatekeeper, which itself is registered with prefix 31 at the 
ViDeNet root gatekeepers. The gateway is registered as a service at the office 
gatekeeper (with the prefix 5) and connected to the PBX. In the PBX, it is 
configured that all calls to 367 and 167 have to be forwarded to the gateway. In 
today's PBXs this is easily configurable and can often be made available even as 
a web-based service, so users can maintain their own preferences. At the PBX, 
the group number (for making all telephones in a group ring) 501 for the group 
"A" belongs to is also defined. At the gatekeeper the number 500 is configured 
as a backup number, that will be called if the H.323 call is not answered. The 
IP phones are registered with their 1xx number (in this case 167) at the 
CallManager which itself is registered as a gateway serving all these numbers 
(167, 109, 1xx etc) at the gatekeeper. </P>
<P>We can examine the following situations (not a complete list of 
possibilities, but a couple of descriptive ones):</P>
<P>a) a user on the PSTN calls "A" at SURFnet, which has decided to take all 
calls on his H.323 station. When the call for 030-2305367 comes in at the PBX of 
SURFnet, the PBX looks for the terminal (telephone) 67. It recognizes that the 
call has to be forwarded to the gateway. When the call is routed there the H.323 
gatekeeper picks it up and looks for a terminal with number 67, finds it as 
"A"'s H.323 station and forwards the call. The ISDN signalling is done between 
the PBX and the gateway and the call is set up. "A" answers the incoming call on 
his videoconferencing station, only receiving audio, since there is no video 
attached to the signal from the PSTN user. If "A" does not answer on his H.323 
station, then the gatekeeper sees this and dials the backup number 501. The 
gatekeeper recognizes this as a call for the voice service at the gateway 
(prefix 5) and routes the call there. The gatekeeper passes it off to the PBX 
which makes all phones in the group ring. "A" or one of his colleagues can then 
answer the call by picking up any of the phones in the group.</P>
<P>b) a user using an IP phone dials A's phone number. For this example "A" has 
his regular phone registered with 67 at the PBX and his H.323 station as 97 at 
the gatekeeper. The H.323 IP phones (or the Cisco IP phones through the 
Callmanager) when connected to the GDS/ViDeNet system can find each other 
through that hierarchy. If someone using an IP phone dials 0031302305367 then 
the Call manager recognizes this as an international call and forwards it to the 
local gatekeeper. The gatekeeper sees that it is an international call and 
forwards it to the ViDeNet root gatekeeper. Here the prefix 31 is recognized and 
the call is forwarded to the Dutch National gatekeeper. There the prefix 3023053 
is recognized and the call is forwarded to the SURFnet office gatekeeper. Here 
the number 67 is recognized. Not having a station with 67 registered there it 
falls back to forwarding the call to the gateway which routes it to the PBX. 
Here the phone with 67 is found and the call is setup.</P>
<P>c) a videoconferencing station dials A's IP phone. Someone using a H.323 
videoconferencing station finds "A"'s number on a card as 00312305167. He dials 
the number. Like in the example above, through the ViDeNet hierarchy the call 
gets to the SURFnet office gatekeeper who sees that the call is for 167. In its 
tables it finds that that number belongs to the CallManager and routes the call 
there. The CallManager acts as a H.323-Skinny gateway and forwards the call to 
the IPphone.</P>
<P>Note. If "A" had also used the number 030-2305367 for his IP phone he would 
have had to make the choice at the gatekeeper to route all calls to the H.323 VC 
terminal, or to the IP phone, since there cannot be two devices registered with 
the same alias (E.164 no.) at the same gatekeeper.</P>
<P>Local dialing between the systems is supported too: "A" can call 97 from his 
phone to reach his H.323 station, or 167 to reach his IP phone. The other way 
around (from IP phone or H.323 station) he needs to use a defined prefix (5 for 
voice calls, see above), so 5367 will ring his normal PSTN phone.</P>
<P>In case the MCU was involved, people using either a PSTN device, or a H.323 
IP phone, or a videoconferencing station, would dial the routable E.164 number 
of the multipoint conference that is registered at the office gatekeeper, as if 
it was a H.323 terminal.</P>
<P>The next step towards full integration is the introduction of a SIP proxy and 
SIP-H.323 gateway making it possible to extend the range of used devices even 
further.</P>
<P>Note, the example above relies on a numeric dialing plan. Alphanumeric 
dialing and routing is implemented differently, see section 2.xx on 
Addressing.</P></DIV></DIV>
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