23/3/2020 Unified Communication: H.

323 Gateway

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Unified Communication
Wednesday, 15 August 2012

H.323 Gateway

H.323 is a recommenda on from the ITU Telecommunica on Standardiza on Sector (ITU-T) that defines the
protocols to provide audio & video communica on sessions on any packet network. The H.323 Standard
addresses call signaling and control, mul media transport and control, and bandwidth control for point-to-
point and mul -point conferences

H.323 is a system specifica on that describes the use of several ITU-T and IETF protocols. The protocols that
comprise the core of almost any H.323 system
H.225 Call Signaling, which is used between any two H.323 en es in order to establish
communica on.
H.245 control protocol for mul media communica on, which describes the messages and
procedures used for capability exchange, opening and closing logical channels for audio, About Me Blog Archive
video and data, control and indica ons.
Real- me Transport Protocol (RTP), which is used for sending or receiving mul media Muhammad ►
► 2013 (3)
Nawaz (CCIE
informa on (voice, video, or text) between any two en es. ▼
▼ 2012 (8)
VOICE/SEC/SP)
Codecs that are widely implemented by H.323 equipment include: ►
► September
View my complete (3)
Audio codecs: G.711, G.729 (including G.729a), G.723.1, G.726, G.722, G.728 profile
▼
▼ August (5)
Text codecs: T.140
Multiple PRI's
Video codecs: H.261, H.263, H.264
configuratio
n on Voice
gateway
MGCP
Gateway
H.323
Gateway
PRI/E1
Configurati
on on Cisco
Voice
Gateway
Session
Initiation
Protocol
(SIP)

· Peer to Peer protocol

· No central control
· Each gateway act on its own
· All PSTN signaling terminates on gateway
· H.323 and H.245 signaling communica on over TCP between gateway & CUCM (Cisco unified
Communica on Manager )
· Media over UDP directory between gateway and IP phones, CUCM responsible for call setup/tear-
down and nego a on capability only
· Gateway status on CUCM always remain “Unknown”
· Dial plan and transla on can be configured per gateway basis.
· Call preserva on for Cisco SRST

Card type e1 0 0
!
network-clock-participate wic 0
network-clock-select 1 E1 0/0/0
!
isdn switch-type primary-net5
!

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controller E1 0/0/0
framing no-crc4
linecode hdb3
pri-group timeslots 1-31
!
voice-port 0/0:15

!
dial-peer voice 1 pots
Incoming called-number.
direct-inward-dial
port 0/0/0:15

Now we have to check the status of PRI using show isdn status command. If you see the layer 2 state as
MULTIPLE_FRAME_ESTABLISHED this means that PRI is successfully configured and ready to make/receive
Calls. If it doesn’t show the MULTIPLE_FRAME_ESTABLISHED then follow the PRI/E1 blog

VOICE_GW#show isdn status

Global ISDN Switchtype = primary-net5

ISDN Serial0/0/0:15 interface

dsl 0, interface ISDN Switchtype = primary-net5

Layer 1 Status:

ACTIVE

Layer 2 Status:

TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED

Layer 3 Status:

5 Ac ve Layer 3 Call(s)

Ac vated dsl 0 CCBs = 1

CCB:callid=7D5, sapi=0, ces=0, B-chan=9, calltype=DATA

The Free Channel Mask: 0xFFFF78FC

ISDN Serial0/0/0:15 interface

dsl 1, interface ISDN Switchtype = primary-net5

Layer 1 Status:

ACTIVE

Layer 2 Status:

H.225 mer is set to three seconds; the router a empts a connec on to the primary CUCM Server. If it does
not receive a response in three seconds; it falls back to the secondary CUCM Server.
!
voice class h323 1
h225 meout tcp establish 3

! Use this interface specific command, H323-gateway voip bind scraddr x.x.x.x, in the H.323 gateway to force
it to use a specific IP address for H.225 setup messages with CUCM.

interface GigabitEthernet0/0
ip address 10.10.10.100 255.255.255.0
h323-gateway voip interface
h323-gateway voip bind srcaddr 10.10.10.100
!
Note: we assume that PSTN Provider sending 4digit of DID called number if Service Provider is sending more
than 4 digit then transla on rule required to translate the called number into 4 digits.

! Dial peer 11 route the incoming calls from PSTN for 4000-4999 towards the Publisher CUCM.( When you
set the preference order, the lower the preference number, the higher the priority. The highest priority is
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given to the dial peer with preference order 0 and it is the default value. You can have a preference value
between 0 to 10)

dial-peer voice 11 VoIP

Descrip on ******CUCM Pub****
des na on-pa ern 4…
session target ipv4: 10.10.10.2
dtmf-relay h245-alphanumeric
voice-class h323 1
codec g711ulaw
Preference 2
no vad
!
! Dial peer 12 route the incoming calls from PSTN for 4000-4999 towards the Subscriber CUCM

dial-peer voice 12 VoIP

Descrip on ******CUCM Sub****
des na on-pa ern 4...
session target ipv4: 10.10.10.3
dtmf-relay h245-alphanumeric
voice-class h323 1
codec g711ulaw
preference 1
no vad
!

! Below transla on rule will add the prefix (9) in all incoming calling number from PSTN. This will be help
the users to make call using missed/received calls from their phone.
!
voice transla on-rule 1 rule 1 /^\(.*\)/ /9\1/
!
voice transla on-profile addinPrefix
translate calling 1
!
voice-port 0/0/0:15
transla on-profile incoming addinPrefix
!
! Dial Plan for outgoing calls to PSTN

dial-peer voice 991 pots

Descrip on *** 3 digit outgoing call*****
des na on-pa ern 9[1-9]..
port 0/0/0:15
forward-digits 3
!
dial-peer voice 992 pots
Descrip on *** 7 digit outgoing call*****
des na on-pa ern 9[2-9]......
port 0/0/0:15
forward-digits 7
!
dial-peer voice 993 pots
Descrip on *** 10 digit outgoing call*****
des na on-pa ern 905[056].......
port 0/0/0:15
prefix 05
!
dial-peer voice 994 pots
Descrip on *** 9 digit outgoing call*****
des na on-pa ern 90[234679].......
port 0/0/0:15
prefix 0
!
dial-peer voice 995 pots
Descrip on ** interna onal outgoing calls***
des na on-pa ern 900T
port 0/0/0:15
prefix 00
!

H.323 Gateway configura on in Cisco Unified Communica on Manager

Click on Device and then Gateway

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Then click on Add New

Select H.323 Gateway from the Gateway Type list

Device Name: 10.10.10.100 Gateway address and click on save

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Device >> Gateway

Troubleshoo ng

Problem
Calls from a Cisco IP phone to a PSTN/PBX phone ring, but as soon as the called party picks up the phone,
both ends hear a fast-busy

Symptom
There is a CODEC mismatch between the Cisco IP phone and the H.323 gateway.

Solu on
Check these items in CUCM and the IOS® configura on:
1. Double check the Region and Device Pool configura on in CUCM, where CODEC is defined. Newer
Cisco IP phones (79xx) support G.711 and G.729.
2. If G.729 is needed between the gateway and Cisco IP phone, make sure the Media Termina on
Point Required box is not checked on the Gateway Configura on page. Otherwise, the gateway
connec on always uses G.711.
3. Make sure the proper CODEC is defined under voip dial-peer on the H.323 gateway. The default is
G.729r8.

Problem
Inbound calls from PSTN do not complete to CUCM and the Cisco IP phone, while the CUCM and H.323
gateway are properly configured.

Symptom
From debug cch323 h225 on the H.323 gateway, it sends out an H.225 setup message to CUCM, but never
hears back. This is because CUCM does not know how to reach the IP address that the H.323 gateway used
for the H.225 setup message.

Solu on
Use the interface specific command, H323-gateway voip bind scraddr x.x.x.x, in the H.323 gateway to force
it to use a specific IP address (which is reachable by CUCM) to send the H.225 setup message.

Problem

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Inbound calls from the PSTN to CUCM do not work, while outbound calls from CUCM to the PSTN work fine.

Symptom
From debug voip ccapi inout on the H.323 gateway, CUCM disconnects the call because of an unassigned
number (0x1) or invalid number (0x1C).

Solu on
Check the CUCM configura on to make sure that the H.323 gateway is in a Calling Search Space that enables
it to reach the Par ons that the IP phones belong to.

Problem
Inbound calls from the PSTN to CUCM do not work, while outbound calls from CUCM to the PSTN work fine.

Symptom
From debug voip ccapi inout on the H.323 gateway, the gateway disconnects the call because of an
unassigned number (0x1) or invalid number (0x1C).

Solu on
Check the IOS configura on for any number-expansions or transla on pa erns. Any called number that
comes from the PSTN needs to go through these pa erns before it is matched to the VoIP dial-peer.

Posted by Muhammad Nawaz (CCIE VOICE/SEC/SP) at 03:20

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