Wednesday, 15 August 2012

H.323 Gateway

 
H.323 is a recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio & video communication sessions on any packet network. The H.323 Standard addresses call signaling and control, multimedia transport and control, and bandwidth control for point-to-point and multi-point conferences

H.323 is a system specification 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 entities in order to establish communication.
  • H.245 control protocol for multimedia communication, which describes the messages and procedures used for capability exchange, opening and closing logical channels for audio, video and data, control and indications.
  • Real-time Transport Protocol (RTP), which is used for sending or receiving multimedia information (voice, video, or text) between any two entities.
Codecs that are widely implemented by H.323 equipment include:
  • Audio codecs: G.711, G.729 (including G.729a), G.723.1, G.726, G.722, G.728
  • Text codecs: T.140
  • Video codecs: H.261, H.263, H.264



·         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 communication over TCP between gateway & CUCM (Cisco unified Communication Manager )
·         Media over UDP directory between gateway and IP phones, CUCM responsible for call setup/tear-down and negotiation capability only
·         Gateway status on CUCM always remain “Unknown”
·         Dial plan and translation can be configured per gateway basis.
·         Call preservation 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 
!
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 Active Layer 3 Call(s)
    Activated 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 timer is set to three seconds; the router attempts a connection 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 timeout 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 translation 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 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 
 Description ******CUCM Pub**** 
 destination-pattern 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 
 Description ******CUCM Sub**** 
 destination-pattern 4... 
 session target ipv4: 10.10.10.3 
 dtmf-relay h245-alphanumeric
 voice-class h323 1 
  codec g711ulaw
  preference 1
 no vad
!

! Below translation 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 translation-rule 1  rule 1 /^\(.*\)/ /9\1/
!
voice translation-profile addinPrefix 
 translate calling 1
 !
voice-port 0/0/0:15
translation-profile incoming addinPrefix
!
! Dial Plan for outgoing calls to PSTN

dial-peer voice 991 pots 
 Description *** 3 digit outgoing call***** 
 destination-pattern 9[1-9].. 
  port 0/0/0:15 
 forward-digits 3
!
dial-peer voice 992 pots 
 Description *** 7 digit outgoing call***** 
 destination-pattern 9[2-9]...... 
 port 0/0/0:15 
 forward-digits 7
!
dial-peer voice 993 pots
Description *** 10 digit outgoing call*****
 destination-pattern 905[056].......
 port 0/0/0:15
 prefix 05
!
dial-peer voice 994 pots
 Description *** 9 digit outgoing call*****
 destination-pattern 90[234679].......
 port 0/0/0:15
 prefix 0
!
dial-peer voice 995 pots
 Description ** international outgoing calls***
 destination-pattern 900T
 port 0/0/0:15
 prefix 00
!

H.323 Gateway configuration in Cisco Unified Communication Manager
Click on Device and then Gateway


 







 

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



Device >> Gateway







Troubleshooting

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.

Solution
 Check these items in CUCM and the IOS® configuration:
1.       Double check the Region and Device Pool configuration 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 Termination Point Required box is not checked on the Gateway Configuration page. Otherwise, the gateway connection 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.

Solution
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
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).

Solution
Check the CUCM configuration to make sure that the H.323 gateway is in a Calling Search Space that enables it to reach the Partitions 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).

Solution
Check the IOS configuration for any number-expansions or translation patterns. Any called number that comes from the PSTN needs to go through these patterns before it is matched to the VoIP dial-peer.

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