Saturday 7 January 2012

Voice gateway protocols


H225 setup:ringing

H245 feature:negotiate codec g711,g729

H225 RAS: communicate to gatekeeper



        RRQ            RRQ

gateway-----gatekeeper-----gateway

        RCF            RCF

once call finished:disengage confirm



gatekeeper(control center):bandwidth,where to forward the call, allow the call forward/drop





gatekeeper commands:

zone local

zone remote

zone prefix

no shut



gateway commands:

h323-gateway voip int

h323-gateway voip ID

h323-gateway voip h323-ID

h323-gateway voip bind

session target RAS



gateKeeper:

#gatekeeper

#zone local gk1.cisco.com cisco.com 10.1.1.1

#zone remote gk2.cisco.com cisco.com 10.1.1.2

#zone prefix gk1.cisco.com 7...

#zone prefix gk2.cisco.com 8...

#no shut

#exit



gateWay

#interface f0/0

#h323-gateway voip interface

#h323-gateway voip id gk1.cisco.com ipaddr 10.1.1.1 1718

#h323-gateway voip h323-id myrouter

#h323-gateway voip bind srcaddr 1.1.1.1

#exit

#dial-peer voice 100 voip

#destination-pattern ....

#session target ras (gatekeeper)

#exit



MGCP relys on call agent(CUCM)



events: on-hook,off-hook,hookflash,fax/modem tones,dial DTMF digits

signal: busy tone,call waiting tone,dial tone,DTMF tone,ringing tone,ringback tone



configuration:

#mgcp

#mgcp call-agent 192.168.12.120

#dial-peer voice 1 pots

#port 1/1/0

#application MGCPAPP

#exit





SIP concepts and configuration



1.the SIP standards and architechture

SIP was created by IETF

based on many of the previous protocols(HTTP,SMTP,DNS..)

uses text-based(ASCII) communication

more of an "all-in-one"



2.the SIP components

database-->SIP proxy-->user agent



3.SIP request/response messages

invite->ACK->BYE->cancel->options->register->info



4.SIP addressing

using E.164 address

sip://1212@cisco.com

using FQDNS

SIP://123@cisco.com



5.sample SIP call setup

invite(SDP)->trying-->ringing(SDP)-->OK(400)-->(ACK)-->RTP-->RTCP





6.configuration a cisco router for SIP

#conf t

#sip-ua

#sip-server dns:sip123(ipv4:10.1.1.1)

#exit

to a sip-server

#dial-peer voice 15 voip

#destination-pattern ....

#session protocol sipv2

#session target sip-server

to a normal peer router

#dial-peer voice 16 voip

#destination-pattern 99...

#sessin protocol sipv2

#session target ipv4:5.5.5.5

 H323 VS MGCP link:

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