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Meddane
VIP
VIP

Dial Peers for Internal Calls

 

Meddane_0-1683753486317.png

From Webex Calling to CUCM

The Voice Class Tenant 200 is a trunk to process inbound and outbound calls from and to Webex Calling and this will be associated to the Dial Peer Voice 201 with voice-class sip tenant 200 command. Specify the webex servers using the sip-server command.  We will use the session target sip-server command under each outgoing dial peers to Webex Calling.

 

voice class tenant 200

registrar dns:40462196.cisco-bcld.com scheme sips expires 240 refresh-ratio 50 tcp tls

credentials number test8789_LGU username test3350_LGU password 0 bjljJ2VQji realm

BroadWorks

authentication username test3350_LGU password 0 bjljJ2VQji realm BroadWorks

authentication username test3350_LGU password 0 bjljJ2VQji realm 40462196.cisco-bcld.com

no remote-party-id

sip-server dns:40462196.cisco-bcld.com

connection-reuse

srtp-crypto 200

session transport tcp tls

url sips

error-passthru

asserted-id pai

bind control source-interface GigabitEthernet1

bind media source-interface GigabitEthernet1

no pass-thru content custom-sdp

sip-profiles 200

outbound-proxy dns:la01.sipconnect-us10.cisco-bcld.com

privacy-policy passthru

 

We need to define the pattern to uniquely identify the specific organization to be matched by the incoming dial peer in the Local Gateway from Webex Calling.

Incoming calls from Webex Calling are identified with voice class URI. The SIP INVITE Request-URI sent from the Webex Calling cloud to the local gateway contains a dtg parameter that is generated in Control Hub.

 

voice class uri WBX-URI sip

pattern dtg=test3350.lgu

 

Configure the Dial Peer Group 300 to point to outbound Dial Peer Voice 301. The purpose is to route incoming calls matching the inbound Dial Peer 201 from Webex Calling to Dial Peer 301.

 

voice class dpg 300

description Incoming calls webex (Dial Peer 201) to CUCM (Dial Peer 301)

dial-peer 301 preference 1

 

The call from Webex Calling uses the dial peer 201 because there is a match using the incoming uri request WBX-URI command, After matching the incoming calls from Webex Calling, they are routed to the dial peer 301 because the destination dpg 300 command.

 

dial-peer voice 201 voip

description Inbound/Outbound to and from Webex Calling

max-conn 250

destination-pattern BAD.BAD

session protocol sipv2

session target sip-server

destination dpg 300

incoming uri request WBX-URI

voice-class codec 99

voice-class stun-usage 200

no voice-class sip localhost

voice-class sip tenant 200

dtmf-relay rtp-nte

srtp

no vad

 

The voice class tenant 100 is a trunk to the CUCM. It will be applied on all Outbound dial peers facing the CUCM.

 

voice class tenant 100

session transport udp

url sip

error-passthru

bind control source-interface GigabitEthernet2

bind media source-interface GigabitEthernet2

no pass-thru content custom-sdp

 

Define the CUCM as the target for the dial peer 301. In order to do that, you need to create voice class server group with all the IP addresses of the CUCM nodes with the callmanager service enabled.

 

voice class server-group 301

ipv4 10.1.5.15

Because the Dial Peer Group 300 configured previously points to Dial Peer Voice 301 with dial-peer 301 preference 1 command the call is routed to dial peer 301. This Dial Peer points to IP of CUCM using the voice class server-group command.

 

dial-peer voice 301 voip

description Outgoing dial-peer to CUCM

destination-pattern BAD.BAD

session protocol sipv2

session server-group 301

voice-class codec 99

dtmf-relay rtp-nte

voice-class sip tenant 100

no vad

 

From CUCM to Webex Calling

Configure the Voice Class Tenant 300 that will be applied to Inbound dial peer 300 from the CUCM.

It will be associated to Dial Peer Voice 300 using the voice-class sip tenant 300 command.

 

voice class tenant 300

bind control source-interface GigabitEthernet2

bind media source-interface GigabitEthernet2

no pass-thru content custom-sdp

 

Configure the voice class URI to match the CUCM IP address.

 

voice class uri 300 sip

host ipv4:10.1.5.15

 

Define the dial peer group 200, the purpose is to route the calls to dial peer 201.

Configure the Dial Peer Group 200 to point to outbound Dial Peer Voice 201. The purpose is to route incoming calls matching the inbound Dial Peer 300 from Webex Calling to Dial Peer 201.

 

voice class dpg 200

description Incoming CUCM (Dial Peer 300) to Webex Calling (Dial Peer 201)

dial-peer 201 preference 1

 

The dial peer 300 matches the CUCM IP address based on the voice class URI using the incoming uri via 300 command, then the call is routed to the dial peer group 200 because the destination dpg 200 command.

 

dial-peer voice 300 voip

description Incoming dial-peer from Unified CM to Webex

session protocol sipv2

destination dpg 200

incoming uri via 300

voice-class codec 99

dtmf-relay rtp-nte

voice-class sip tenant 300

 

Because the Dial Peer Group 200 points to Dial Peer Voice 201, the call is routed to dial peer 201.

The dial peer 201 configured previously for inbound calls to Webex Calling will be used as the outbound Dial Peer for incoming calls from PSTN.

Because the Dial Peer Group 200 points to Dial Peer Voice 201 the call is routed to dial peer 201, the session target sip-server command under the Dial Peer 201 instructs the CUBE to use the sip-server that points to Webex Servers configured previously in voice class tenant 200 to send outgoing calls to Webex Calling.

 

dial-peer voice 201 voip

description Inbound/Outbound Webex Calling

max-conn 250

destination-pattern BAD.BAD

session protocol sipv2

session target sip-server

destination dpg 300

incoming uri request WBX-URI

voice-class codec 99

voice-class stun-usage 200

no voice-class sip localhost

voice-class sip tenant 200

dtmf-relay rtp-nte

srtp

no vad

Dial Peers for PSTN Calls

 

PSTN WBX.png

 

Configure the Voice Class Tenant 200. This is a trunk to process inbound and outbound calls from and to Webex Calling and this will be associated to the Dial Peer Voice 2001 with voice-class sip tenant 200 command. The Dial Peer Voice 2001 will be used for inbound and outbound to and from Webex Calling. Specify the webex servers using the sip-server command.  We will use the session target sip-server command under each outgoing dial peers to Webex Calling.

 

voice class tenant 200

registrar dns:40462196.cisco-bcld.com scheme sips expires 240 refresh-ratio 50 tcp tls

credentials number test8789_LGU username test3350_LGU password 0 bjljJ2VQji realm

BroadWorks

authentication username test3350_LGU password 0 bjljJ2VQji realm BroadWorks

authentication username test3350_LGU password 0 bjljJ2VQji realm 40462196.cisco-bcld.com

no remote-party-id

sip-server dns:40462196.cisco-bcld.com

connection-reuse

srtp-crypto 200

session transport tcp tls

url sips

error-passthru

asserted-id pai

bind control source-interface GigabitEthernet1

bind media source-interface GigabitEthernet1

no pass-thru content custom-sdp

sip-profiles 200

outbound-proxy dns:la01.sipconnect-us10.cisco-bcld.com

privacy-policy passthru

 

Configure the voice class tenant 400, this is a trunk to the ITSP. It will be applied on Outbound Dial Peer 101 facing the PSTN.

 

It will be associated to the inbound Dial Peer Voice 101 using the voice-class sip tenant 400 command.

 

voice class tenant 400

session transport udp

url sip

error-passthru

bind control source-interface GigabitEthernet2

bind media source-interface GigabitEthernet2

 

Configure the Voice Class Tenant 500 that will be applied on all Inbound dial peers from the PSTN.

It will be associated to the inbound Dial Peer Voice 100 using the voice-class sip tenant 500 command.

 

voice class tenant 500

bind control source-interface GigabitEthernet2

bind media source-interface GigabitEthernet2

no pass-thru content custom-sdp

 

Configure the voice class URI to match the ITSP PSTN IP address.

 

voice class uri 100 sip

host ipv4:198.12.13.2

 

We need to define the pattern to uniquely identify the specific organization to be matched by the incoming dial peer in the Local Gateway from Webex Calling.

Incoming calls from Webex Calling are identified with voice class URI. The SIP INVITE Request-URI sent from the Webex Calling cloud to the local gateway contains a dtg parameter that is generated in Control Hub.

 

voice class uri 200 sip

pattern dtg=test3350.lgu

 

Configure a Dial Peer Group for calls from Webex to PSTN.

 

Configure the Dial Peer Group 100 to point to outbound Dial Peer Voice 101. The purpose is to route incoming calls matching the inbound Dial Peer 2001 from Webex Calling to Dial Peer 101.

 

voice class dpg 100

description Incoming WBX (Dial Peer 200) to PSTN (Dial Peer 101)

dial-peer 101 preference 1

 

Configure a Dial Peer Group for calls from PSTN to Webex.

 

Configure the Dial Peer Group 400 to point to outbound Dial Peer Voice 2001. The purpose is to route incoming calls matching the inbound Dial Peer 100 from ITSP PSTN to Dial Peer 2001.

 

voice class dpg 400

description Incoming PSTN (Dial Peer 100) to WBX (Dial Peer 201)

dial-peer 2001 preference 1

 

Configure an outbound Dial Peer to PSTN.

 

Because the Dial Peer Group 100 configured previously points to Dial Peer Voice 101 with dial-peer 101 preference 1 command the call is routed to dial peer 2001. This Dial Peer points to IP of ITSP using the session target ipv4 :198.18.133.3 command.

 

dial-peer voice 101 voip

description Outgoing dial-peer to PSTN

destination-pattern BAD.BAD

session protocol sipv2

session target ipv4: 198.12.13.2

voice-class codec 99

voice-class sip tenant 400

dtmf-relay rtp-nte

no vad

 

Configure an outbound Dial Peer to Webex Calling.

Because the Dial Peer Group 400 points to Dial Peer Voice 2001 the call is routed to dial peer 2001, the session target sip-server command under the Dial Peer 2001 instructs the CUBE to use the sip-server that points to Webex Servers configured previously in voice class tenant 200 to send outgoing calls to Webex Calling.

 

dial-peer voice 2001 voip

description Inbound/Outbound to and from Webex Calling

destination-pattern BAD.BAD

session protocol sipv2

session target sip-server

voice-class codec 99

voice-class stun-usage 200

no voice-class sip localhost

voice-class sip tenant 200

dtmf-relay rtp-nte

srtp

no vad

 

Configure inbound Dial Peer for Incoming Calls from PSTN

Configure inbound dial peer 100 matches the ITSP IP address based on the voice class URI 100 using the incoming uri via 100 command, then the incoming call is routed to the dial peer group 400 because the destination dpg 400 command.

 

dial-peer voice 100 voip

description Incoming dial-peer from PSTN

session protocol sipv2

destination dpg 400

incoming uri via 100

voice-class codec 99

voice-class sip tenant 500

dtmf-relay rtp-nte

no vad

 

Configure inbound Dial Peer for Incoming Calls from Webex Calling

Because the Dial Peer Group 100 confgured previously points to Dial Peer Voice 101, the call is routed to the outgoing dial peer 101.

The dial peer 2001 configured previously for outbound calls to Webex Calling will be used as the inbound Dial Peer for incoming calls from Webex Calling.

As said previously, incoming calls from Webex Calling are identified with voice class URI 200. The SIP INVITE Request-URI sent from the Webex Calling cloud to the local gateway contains a dtg parameter that is generated in Control Hub, therefore configure the Dial Peer 201 with the incoming uri via 200 command.

After matching the incoming calls from Webex Calling, they are routed to the dial peer 2001 because the destination dpg 400 command.

 

dial-peer voice 2001 voip

description Inbound/Outbound to and from Webex Calling

destination-pattern BAD.BAD

session protocol sipv2

session target sip-server

voice-class codec 99

voice-class stun-usage 200

no voice-class sip localhost

voice-class sip tenant 200

dtmf-relay rtp-nte

destination dpg 100

incoming uri request 200

srtp

no vad

 

 

Comments

In general a great article. If I may suggest one alteration it would be to not use the same dial peer for both in and outbound calls for the call leg to/from Webex as that makes it much harder to follow the flow of the call in the SBC and that can down the line delay any troubleshooting efforts. Also as you from what I can tell do not use the same for any other call leg it creates a difference in configuration that in it self can be confusing.

One other small suggestion is to use none numeric names for your URI classes as it’s much easier to know what the intended use is if you name them with an name like WEBEX or CUCM.

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