Huawei AR1200 Series Configuration Manual - Voice page 9

Huawei AR1200 Series Enterprise Routers
Configuration Guide - Voice
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Figure 1-1 SIP AG as a voice gateway
POTS
SIP AG call flow
The SIP AG call flow is as follows:
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Issue 03 (2011-09-30)
A user connects to the SIP AG through the PSTN. The SIP AG converts analog signals into
digital signals and compresses and packetizes the digital signals so that the voice packets
can be transmitted on the IP network.
Voice packets are transmitted to the SIP AG through the IP network. The IP voice gateway
of the called party converts the voice packets into analog voice signals and transmits them
to the terminal of the called party through the PSTN.
SIPAG Eth1/0/0
Modem FAX
A user picks up the phone and the SIP AG detects offhook signals.
The DSP module of the SIP AG generates a dial tone.
The user hears the dial tone played by the session application of the SIP AG and starts to
dial up.
The session application of the SIP AG collects the number dialed by the user.
The session application of the SIP AG matches the number with preconfigured called
number profiles.
When the voice gateway of the calling party successfully matches the number with a
preconfigured called number profile, the number is mapped to the voice gateway of the
called party.
The SIP AG initiates a call over the IMS and establishes a logical channel for each call.
Then the channel is used to send and receive voice data.
The IMS searches for the destination phone and initiates a call.
After the called party picks up the phone, a call is set up. After the calling party or called
party hangs up the phone, the call is ended.
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