- The router has private (Gi0/0) and public (Gi0/1) interfaces
- Outbound calls are prefixed with a 9
- SIP-UA.com DIDS are 2145556040 – 2145556069
- SIP-UA.com Username is 100001 and password is 1357924680
Configure DNS for sip-ua proxy lookup. We use SRV records for redundancy, so configuring DNS is necessary as part of our redundancy plan
ip domain-lookup ip name-server 18.104.22.168
Global Voice Parameters
Use the following commands to enable cube, allow SIP traffic from trusted sources and setup ‘address-hiding’
voice service voip allow-connections sip to sip ip address trusted list ipv4 22.214.171.124 255.255.255.240 ipv4 10.10.10.0 255.255.255.0 ipv4 126.96.36.199 255.255.255.255 address-hiding sip ! !
SIP-UA.com Registration & Authorization Settings
sip-ua credentials username 100001 password 1357924680 realm sip-ua.com authentication username 100001 password 1357924680 registrar dns:proxy.sip-ua.com expires 60 sip-server dns:proxy.sip-ua.com retry invite 2 timers trying 150
The following translation rules will strip the prefix of 9 from outbound calls and set the caller-id of outbound calls to 2125556040.
voice translation-rule 1 rule 1 /^9/ // voice translation-rule 2 rule 1 /^.*$/ /2125556040/ voice translation-profile SIP-UA translate called 1 translate calling 2
Outbound SIP-UA.com Dial-Peer
The following dial-peer is for all 11-digit and 10-digit calls starting with 9. Configure the g711ulaw as the codec and bind the source interface to the public Gi0/1 interface
dial-peer voice 1 voip desc ** Outbound Dial-Peer to SIP-UA.com ** translation-profile outgoing SIP-UA session protocol sipv2 session target sip-server destination-pattern 91?[2-9]..[2-9]……$ codec g711ulaw voice-class sip early-offer forced voice-class sip bind control source-interface Gi0/1 voice-class sip bind media source-interface Gi0/1 dtmf-relay rtp-nte no vad
Inbound SIP-UA.com Dial-Peer
dial-peer voice 2 voip desc ** Incoming Dial-Peer from SIP-UA.com ** session protocol sipv2 session target sip-server incoming called-number 21455560. codec g711ulaw voice-class sip early-offer forced voice-class sip bind control source-interface Gi0/1 voice-class sip bind media source-interface Gi0/1 dtmf-relay rtp-nte no vad
Outbound CUCM Dial-Peers
dial-peer voice 3 voip description ** Subscriber ** destination-pattern 21455560. session protocol sipv2 session target ipv4:10.10.10.11 codec g711ulaw voice-class sip bind control source-interface Gi0/0 voice-class sip bind media source-interface Gi0/0 dtmf-relay rtp-nte no vad dial-peer voice 4 voip description ** Publisher ** preference 1 destination-pattern 21455560. session protocol sipv2 session target ipv4:10.10.10.10 codec g711ulaw voice-class sip bind control source-interface Gi0/0 voice-class sip bind media source-interface Gi0/0 dtmf-relay rtp-nte no vad
Inbound CUCM Dial-Peer
dial-peer voice 5 voip incoming called-number 91?[2-9]..[2-9]……$ session protocol sipv2 codec g711ulaw voice-class sip bind control source-interface Gi0/0 voice-class sip bind media source-interface Gi0/0 dtmf-relay rtp-nte no vad
Configuring H.323-to-SIP Interworking
Figure 9-16 shows a sample scenario used to configure H.323-to-SIP interworking with a Cisco UBE router. The Cisco Unified Communications Manager cluster in San Jose routes calls to the SIP carrier via a Cisco UBE router. The connection between the Cisco Unified Communications Manager and the Cisco UBE router is H.323 and the connection between the Cisco UBE router and the SIP carrier is SIP.
Figure 9-16 H.323-to-SIP Interworking Scenario
You can follow these steps to configure H.323-to-SIP interworking: Step 1. Enable H.323-to-SIP interworking.
Step 2. Configure H.323 and SIP dial peers to route international calls between the Cisco Unified Communications Manager cluster and the SIP carrier.
Step 1: Enabling H.323-to-SIP Interworking
As with an H.323-to-H.323 connection, by default a Cisco IOS gateway will not allow connections between an H.323 and a SIP VoIP dial peer. To change this behavior and allow H.323-to-SIP connections, use the allow-connections h323 to sip command in voice service configuration mode. Then issue the allow-connections sip to h323 command to enable SIP to H.323 calls, as demonstrated in Example 9-4.
Example 9-4 H.323-to-SIP Interworking
Step 2: Configuring Dial Peers
For a SIP (rtp-nte)-to-H.323 (h245-alphanumeric) call via a Cisco UBE router, if any RTP named telephony event (NTE) packets are sent before the H.323 endpoint answers the call, the DTMF signal is not heard on a terminating gateway (TGW).
Note debug output reveals that the H245 out-of-band messages are sent to the TGW. However, the digits are not heard on the phone.
To avoid sending both in-band and out-of-band tones to the outgoing leg when sending Cisco UBE calls in-band (rtp-nte) to out-of-band (h245-alphanumeric), configure the dtmf-relay rtp-nte digit-drop command on the incoming SIP dial peer. On the H.323 side, configure either dtmf-relay h245-alphanumeric or dtmf-relay h245-signal. This can also be used for H.323-to-SIP calls. Example 9-5 illustrates this dial-peer configuration.
Example 9-5 Dial-Peer Configuration
Media Flow and Transparent Codec Commands
To configure media flow-through or media flow-around, use the following media command:
Note that this command can be issued in dial-peer configuration mode or globally under the voice service configuration mode. The default is media flow-through.
To configure transparent codec pass-through, use the following codec transparent command:
Note that this command can be issued in dial-peer configuration mode or via a codec class.
Configuring Transparent Codec Pass-Through and Media Flow-Around
Figure 9-17 shows a sample scenario used to configure H.323-to-H.323 interworking, including transparent codec pass-through and media flow-around, using a Cisco UBE router. The Cisco Unified Communications Manager cluster in San Jose is connected with the Cisco Unified Communications Manager Express router in Chicago using a Cisco UBE router. Codec negotiation is performed directly between the Cisco Unified Communications Manager cluster and the Cisco Unified Communications Manager Express router, and RTP streams flow directly between the endpoints.
Figure 9-17 Transparent Codec Pass-Through and Media Flow-Around Example Topology
Codec transparency enables a Cisco UBE router to pass codec capabilities between end-points. If you configure transparency, a Cisco UBE router uses the codec that was specified by the endpoints for setting up a call. To enable endpoint-to-endpoint codec negotiation without a Cisco UBE router, use the codec transparent command.
With the default configuration, a Cisco UBE router receives media packets from the inbound call leg, terminates them, and then reoriginates the media stream on an outbound call leg. Media flow-around enables media packets to be passed directly between the endpoints, without the intervention of a Cisco UBE router. The Cisco UBE router continues to handle routing and billing functions. Media flow-around for SIP-to-SIP calls is not supported. Use the media flow-around command to enable media flow-around. Example 9-6 illustrates the use of both the codec transparent and the media flow-around commands.
Example 9-6 Transparent Codec Pass-Through and Media Flow-Around Configuration
Figure 9-18 shows a sample scenario used to configure a Cisco UBE router and a via-zone gatekeeper. A gatekeeper is configured with two standard local zones: San Jose (SJC) and Chicago (CHI). The Cisco Unified Communications Manager Express Router1 is registered in the SJC zone, and the Cisco Unified Communications Manager Express Router3 is registered in CHI zone. Calls between Chicago and San Jose should be routed by the gatekeeper. Instead of routing calls directly between the two zones, the gatekeeper should route the calls through the VIA, which includes a Cisco UBE router.