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CCIE Collaboration Lecture Chapter 4.4 voice gateway cucm sip overview

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Faisal Khan
Faisal Khan

CCIE Collaboration Bootcamp is designed to be a challenging five-day course for CCIE Collaboration candidates ready for CCIE Collaboration Lab Exam. This Bootcamp is designed for CCIE Collaboration candidates in the last months or weeks before their CCIE Collaboration Lab Exam. During the week students will tackle challenging full-day mock labs Monday through Thursday. Candidate will practice strategy, time management, learn test taking strategies and expose any weaknesses in order to resolve them before the lab exam. On the final day of the course

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1 b & www.uccollaborations.com
222 CCNP Voice Plus Overview Voice Gateways & Signaling Protocol. SIP IOS Gateway
333 Overview Session Initiation Protocol (SIP) is an ASCII-based Application-layer control protocol that can be used to establish, maintain, and terminate calls between two or more endpoints SIP is a peer-to-peer protocol SIP provides the following capabilities Determines the location of the target endpoint Determines the media capabilities of the target endpoint Determines the availability of the target endpoint Establishes a session between the originating and target endpoints Handles the transfer and termination of calls Users in a SIP network are identified by unique SIP addresses A SIP address is similar to an e-mail address and is in the format of sip:fkhan@uccollaborations.com
444 SIP Components The peers in a session are called user agents (UAs) User-agent client (UAC)A client application that initiates the SIP request User-agent server (UAS)A server application that contacts the user when a SIP request is received and that returns a response on behalf of the user SIP endpoint is capable of functioning as both a UAC and a UAS but functions only as one or the other per transaction
555 SIP Clients PhonesCan act as either UAS or UAC Softphones Ephone (such as hardware phone) Gateways
666 SIP Servers Proxy server Receives SIP requests from a client and forwards them on the client's behalf Provide call routing Authentication, authorization, network access control Redirect server Provides the client with information about the next hop or hops that a message should take and then the client contacts the next- hop server or UAS directly Registrar server Processes requests from UACs for registration of their current location
777 How SIP Works with a Proxy Server Caller UA sends an INVITE request to the proxy server and then the proxy server determines the path and forwards the request to the called party The called UA responds to the proxy server, which then forwards the response to the caller Request Response
888 How SIP Works with a Redirect Server Caller UA sends a SIP INVITE request to the redirect server Redirect contacts the location server to determine the path to the called party and sends that information back to the caller UA The caller UA then sends a SIP INVITE request directly to the device indicated in the redirect information
999 Perform Basic SIP Configuration How to Configure SIP Bind Features SIP signaling and media paths can advertise the same source IP address on the gateway for certain applications the bind command adds flexibility by assigning a gateway to a specific interface (and therefore the associated address) for the signaling or media application You can obtain a predefined and separate interface for both signaling and media traffic the bind command allows you to configure the source IP address of signaling and media packets to a specific interface's IP address
101010 Enabling Blind Command Global Command voice service voip Sip Bind all source-interface loopback 0 Dial Peer Level dial-peer voice 1 voip session protocol sipv2 voice-class sip bind control source-interface Loopback0 voice-class sip bind media source-interface Loopback0
111111 Configuring SIP Call Transfer Using the Refer Method dial-peer voice 3001 voip application session incoming called-number 1408T destination-pattern 1408T session protocol sipv2 session target ipv4:172.16.25.21 codec g711ulaw ! dial-peer voice 3002 pots application session destination-pattern 9[2-9].. direct-inward-dial port 2/0:0 ! sip-ua retry bye 1 retry refer 3 timers notify 400 timers refer 567 no oli sip-server ipv4:172.16.21.21 SIP Proxy Server voice service voip sip bind control source-interface Loopback0 bind media source-interface Loopback0 !
121212 Configure Gateway to register to SIP Proxy Server To register a Cisco IOS Gateway using username/password sip-ua credentials username fkhan password 7 030752180500 realm voicebootcamp.com registrar ipv4:172.16.21.21 expires 3600 sip-server ipv4:172.16.21.21 ! dial-peer voice 1 voip destination-pattern 011T voice-class codec 1 voice-class sip dtmf-relay force rtp-nte session protocol sipv2 session target sip-server incoming called-number .% dtmf-relay rtp-nte no vad voice service voip redirect ip2ip sip bind control source-interface Loopback0 bind media source-interface Loopback0 registrar server expires max 3600 min 3600 !
131313 Configure Cisco IP Gateway using IP Address dial-peer voice 1 voip destination-pattern 011T voice-class codec 1 voice-class sip dtmf-relay force rtp-nte session protocol sipv2 session target ipv4:142.11.64.21 incoming called-number .% dtmf-relay rtp-nte no vad voice service voip redirect ip2ip sip bind control source-interface Loopback0 bind media source-interface Loopback0 registrar server expires max 3600 min 3600 !
141414 The End

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CCIE Collaboration Lecture Chapter 4.4 voice gateway cucm sip overview

  • 2. 222 CCNP Voice Plus Overview Voice Gateways & Signaling Protocol. SIP IOS Gateway
  • 3. 333 Overview Session Initiation Protocol (SIP) is an ASCII-based Application-layer control protocol that can be used to establish, maintain, and terminate calls between two or more endpoints SIP is a peer-to-peer protocol SIP provides the following capabilities Determines the location of the target endpoint Determines the media capabilities of the target endpoint Determines the availability of the target endpoint Establishes a session between the originating and target endpoints Handles the transfer and termination of calls Users in a SIP network are identified by unique SIP addresses A SIP address is similar to an e-mail address and is in the format of sip:fkhan@uccollaborations.com
  • 4. 444 SIP Components The peers in a session are called user agents (UAs) User-agent client (UAC)A client application that initiates the SIP request User-agent server (UAS)A server application that contacts the user when a SIP request is received and that returns a response on behalf of the user SIP endpoint is capable of functioning as both a UAC and a UAS but functions only as one or the other per transaction
  • 5. 555 SIP Clients PhonesCan act as either UAS or UAC Softphones Ephone (such as hardware phone) Gateways
  • 6. 666 SIP Servers Proxy server Receives SIP requests from a client and forwards them on the client's behalf Provide call routing Authentication, authorization, network access control Redirect server Provides the client with information about the next hop or hops that a message should take and then the client contacts the next- hop server or UAS directly Registrar server Processes requests from UACs for registration of their current location
  • 7. 777 How SIP Works with a Proxy Server Caller UA sends an INVITE request to the proxy server and then the proxy server determines the path and forwards the request to the called party The called UA responds to the proxy server, which then forwards the response to the caller Request Response
  • 8. 888 How SIP Works with a Redirect Server Caller UA sends a SIP INVITE request to the redirect server Redirect contacts the location server to determine the path to the called party and sends that information back to the caller UA The caller UA then sends a SIP INVITE request directly to the device indicated in the redirect information
  • 9. 999 Perform Basic SIP Configuration How to Configure SIP Bind Features SIP signaling and media paths can advertise the same source IP address on the gateway for certain applications the bind command adds flexibility by assigning a gateway to a specific interface (and therefore the associated address) for the signaling or media application You can obtain a predefined and separate interface for both signaling and media traffic the bind command allows you to configure the source IP address of signaling and media packets to a specific interface's IP address
  • 10. 101010 Enabling Blind Command Global Command voice service voip Sip Bind all source-interface loopback 0 Dial Peer Level dial-peer voice 1 voip session protocol sipv2 voice-class sip bind control source-interface Loopback0 voice-class sip bind media source-interface Loopback0
  • 11. 111111 Configuring SIP Call Transfer Using the Refer Method dial-peer voice 3001 voip application session incoming called-number 1408T destination-pattern 1408T session protocol sipv2 session target ipv4:172.16.25.21 codec g711ulaw ! dial-peer voice 3002 pots application session destination-pattern 9[2-9].. direct-inward-dial port 2/0:0 ! sip-ua retry bye 1 retry refer 3 timers notify 400 timers refer 567 no oli sip-server ipv4:172.16.21.21 SIP Proxy Server voice service voip sip bind control source-interface Loopback0 bind media source-interface Loopback0 !
  • 12. 121212 Configure Gateway to register to SIP Proxy Server To register a Cisco IOS Gateway using username/password sip-ua credentials username fkhan password 7 030752180500 realm voicebootcamp.com registrar ipv4:172.16.21.21 expires 3600 sip-server ipv4:172.16.21.21 ! dial-peer voice 1 voip destination-pattern 011T voice-class codec 1 voice-class sip dtmf-relay force rtp-nte session protocol sipv2 session target sip-server incoming called-number .% dtmf-relay rtp-nte no vad voice service voip redirect ip2ip sip bind control source-interface Loopback0 bind media source-interface Loopback0 registrar server expires max 3600 min 3600 !
  • 13. 131313 Configure Cisco IP Gateway using IP Address dial-peer voice 1 voip destination-pattern 011T voice-class codec 1 voice-class sip dtmf-relay force rtp-nte session protocol sipv2 session target ipv4:142.11.64.21 incoming called-number .% dtmf-relay rtp-nte no vad voice service voip redirect ip2ip sip bind control source-interface Loopback0 bind media source-interface Loopback0 registrar server expires max 3600 min 3600 !