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Voice over Internet Protocol (VoIP) currently implements ITU's H.323 specification within Internet Telephony Gateways (ITGs) to signal voice call setup. Session Initiation Protocol (SIP) is a new protocol developed by the Internet Engineering Task Force (IETF) Multiparty Multimedia Session Control (MMUSIC) Working Group as an alternative to H.323. SIP features are compliant with IETF RFC 2543, published in March 1999.
The Cisco SIP functionality equips the Cisco AS5300 access server, and the Cisco 2600 and Cisco 3600 series routers to signal the setup of voice and multimedia calls over IP networks; therefore, the SIP feature, introduced in Cisco IOS Release 12.1(1)T, provides an alternative to H.323 within the VoIP internetworking software. The SIP feature also provides nonproprietary advantages in the areas of:
This document includes the following sections:
The SIP feature includes the following functionality:
The SIP feature meets the needs of service providers that use SIP on the gateways of their VoIP network to:
Although SIP is simpler than H.323, SIP provides similar capabilities in:
Ensure that your access platform has 16 MB Flash and 64 MB DRAM memory minimum, and that I/O memory is set to either 8 or 16 MB.
The SIP feature is dependent on the interoperability of Service Provider Features for VoIP.
Currently you must know the exact address of a server to contact it. SRV records enable administrators to use several servers to provide the same service within a single domain. SRV Resource Records (RRs) allow administrators to define primary and backup servers and move services from host-to-host without affecting service.
SIP Implementation on the gateway follows the methods outlined in Appendix D of RFC 2543 and RFC 2052. The retrieved SRV RRs are sorted based on the Priority field. Then, starting from the Highest Priority level, a server is chosen randomly based on the Weightage assigned to it. The target address for the selected server is resolved using DNS A records. If the selected server fails to provide the service, a new server is chosen within the same Priority level, or the next lower Priority level is chosen if higher priority levels have been exhausted. This is repeated until all the priority levels have been exhausted.
No new or modified standards are supported by this feature.
No new or modified MIBs are supported by this feature.
For descriptions of supported MIBs and how to use MIBs, see Cisco's MIB web site on CCO at: http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml.
RFC 2543 is supported by this feature.
To configure SIP functions on the Cisco AS5300, or the Cisco 2600 or Cisco 3600 series router, perform the following tasks:
For more information on SIP configuration, including call flows, refer to the Session Initiation Protocol Call Flows document on CCO.
A terminating gateway that is not configured as an SIP user agent cannot receive incoming SIP calls. The transport command opens the SIP listener port (5060) to receive SIP (a SIP user agent is configured to listen by default).
To configure the terminating gateway, enter the following commands beginning in global configuration mode:
| Command | Purpose | |
|---|---|---|
Step 1 | Router (config) # sip-ua | Enters the SIP user agent (sip-ua) mode to configure SIP-UA related commands. |
Step 2 | Router (config-sip-ua)# transport {udp|tcp}
| Configures the SIP user agent (sip-ua) for SIP signaling messages. The default value is udp. |
Step 3 | Router (config-sip-ua) | Enters the IP address of the SIP server interface. |
Step 4 | Router (config-sip-ua) | Sets time to wait for a response. |
Step 5 | Router (config-sip-ua)# retry invite number | Configures the SIP signaling timers for retry attempts. |
Step 6 | Router (config-sip-ua)# inband-alerting header-string | Specifies an inband-alerting SIP header. |
When you configure a VoIP dial peer, you must add the following commands:
| Command | Purpose | |
|---|---|---|
Step 1 | Router (config-dial-peer-voice) | Enters the dial-peer mode to configure a VoIP dial peer. |
Step 2 | Router (config-dial-peer-voice) | Defines the telephone number associated with this VoIP dial peer. |
Step 3 | Router (config-dial-peer-voice) # session transport {udp|tcp}
| Enters the session transport type for the SIP user agent. |
Step 4 | Router (config-dial-peer-voice) # session protocol sipv2 | Enters the session protocol type. |
Step 5 | Router (config-dial-peer-voice) # | Specifies the dial peer session target to use the global SIP server. |
When you configure a POTS dial peer, you must add the following commands:
| Command | Purpose | |
|---|---|---|
Step 1 | Router (config-dial-peer-voice) | Enters the dial-peer mode to configure a VoIP dial peer. |
Step 2 | Router (config-dial-peer-voice) | Defines the telephone number associated with this POTS dial peer. |
Step 3 | Router (config-dial-peer-voice) | Associates this POTS dial peer with a specific voice port. |
Step 4 | Router (config-dial-peer-voice) # session transport {udp|tcp}
| Enters the session transport type for the SIP user agent. |
Step 5 | Router (config-dial-peer-voice) # session protocol sipv2 | Enters the session protocol type. |
Step 6 | Router (config-dial-peer-voice) # | Specifies the dial peer session target to use the global SIP server. |
Enter the show running configuration command to verify your configuration.
 |
Note All IP addresses and patterns are examples only. |
See samples of screen output displays for running configurations:
The following shows an example of the output that appears when you enter the show run command. Irrelevant modules are omitted.
version 12.0 .
.
. hostname UA-1 .
.
. ip name-server ip-address .
.
. isdn switch-type primary-5ess .
.
. ! The "description" used on the T1 controller will appear in the FROM header ! of the SIP Invite. .
.
. controller T1 2 framing esf clock source line secondary 1 linecode b8zs pri-group timeslots 1-24 description SIP Gateway UA-1; t1-pri controller 2 .
.
. controller T1 3 framing esf linecode b8zs pri-group timeslots 1-24 description SIP Gateway UA-1; t1-pri controller 3 .
.
. voice-port 2:D .
.
. voice-port 3:D .
.
. ! Below are examples using the different SIP targets (dns, ipv4, sip-server): ! on the VOIP dial-peers: .
.
. dial-peer voice 100 pots destination-pattern 9003 port 3:D prefix 9003 .
.
. dial-peer voice 101 voip destination-pattern 9004 session protocol sipv2 session target sip-server .
.
. dial-peer voice 200 pots destination-pattern 97055500.. direct-inward-dial port 3:D prefix 97055500 .
.
. dial-peer voice 201 voip destination-pattern 98055500.. max-redirects 2 session protocol sipv2 session target sip-server codec g711ulaw ip precedence 5 no vad .
.
. dial-peer voice 300 pots destination-pattern 95055500.. direct-inward-dial port 2:D prefix 95055500 .
.
. dial-peer voice 301 voip destination-pattern 96055500.. max-redirects 10 session protocol sipv2 session target ipv4:172.16.1.1 codec g711ulaw ip precedence 5 no vad ! ! SIP User Agent configuration ! sip-ua retry invite 2 retry response 2 retry bye 2 retry cancel 2 sip-server ipv4:172.16.1.2 .
.
. interface Ethernet0 ip address 172.16.1.3 255.255.255.0 no ip directed-broadcast load-interval 30 no keepalive no cdp enable .
.
. interface Serial2:23 no ip address no ip directed-broadcast isdn switch-type primary-5ess isdn protocol-emulate user isdn incoming-voice modem isdn T203 10000 fair-queue 64 256 0 .
.
. interface Serial3:23 no ip address no ip directed-broadcasT isdn switch-type primary-5ess isdn protocol-emulate user isdn incoming-voice modem isdn T203 10000 fair-queue 64 256 0 .
.
. interface FastEthernet0 ip address 172.16.1.4 255.255.255.4 no ip directed-broadcast load-interval 30 duplex auto speed auto .
.
. dialer-list 1 protocol ip permit
The following shows an example of the output that appears when you enter the show run command. Inapplicable modules are omitted.
version 12.0 .
.
. hostname UA-4 .
.
. controller T1 0 framing esf clock source line primary linecode b8zs ds0-group 0 timeslots 1-24 type e&m-fgb dtmf dnis .
.
. controller T1 1 framing esf clock source line secondary 1 linecode b8zs ds0-group 0 timeslots 1-24 type e&m-fgb dtmf dnis .
.
. voice-port 0:0 .
.
. voice-port 1:0 .
.
. voice class codec 100 codec preference 1 g726r16 codec preference 2 g729r8 codec preference 3 g711alaw codec preference 4 g711ulaw .
.
. dial-peer voice 500 pots destination-pattern 92055500.. port 0:0 prefix 92055500 .
.
. dial-peer voice 600 voip incoming called-number 92055500.. session protocol sipv2 voice-class codec 100 no vad .
.
. dial-peer voice 501 pots destination-pattern 94055500.. port 1:0 prefix 94055500 .
.
. dial-peer voice 601 voip incoming called-number 94055500.. session protocol sipv2 voice-class codec 100 no vad .
.
. interface Ethernet0 ip address 172.16.1.1 255.255.255.1 no ip directed-broadcast load-interval 30 .
.
. interface FastEthernet0 ip address 172.16.1.2 255.255.255.2 no ip directed-broadcast load-interval 30 duplex auto speed auto
This section documents new or modified commands. All other commands used with this feature are documented in the Cisco IOS Release12.1(1)T command reference publications.
For more information on the search and filter functionality, refer to the Cisco IOS Release 12.1(1)T feature module titled CLI String Search.
This section documents both new and modified commands.
To reset the value of a command to its default, enter the default SIP user-agent configuration command.
default { inband-alerting | max-forwards | retry {invite | response | bye | cancel } | sip-server | timers { trying | connect | disconnect | expires } | transport }Syntax Description
inband-alerting | Resets inband-alerting to its default of generating the header "Require: com.cisco.inband-alerting" in outgoing INVITE messages. |
max-forwards | Resets max-forwards to its default of 6. |
retry {invite | response | bye | cancel } | Resets the specified retry to its default (6 for invite and response; 10 for bye and cancel). |
sip-server | Resets the sip-server to a null value. |
timers { trying | connect | disconnect | expires } | Resets the specified retry to its default (500 for trying, connect, and disconnect; 180000 for expires). |
transport | Resets transport to the default of both UDP and TCP enabled. |
Defaults
There are no default behaviors or values for this command.
Command Modes
SIP user-agent configuration
Command History
| Release | Modification |
|---|---|
12.1(1)T | This command was introduced. |
Examples
Router (config)# sip-ua Router (config-sip-ua)# default inband-alerting
Related Commands
| Command | Description |
|---|---|
Enables the SIP user-agent configuration commands, with which you configure the user agent. |
To exit the SIP user agent configuration, use the exit sip-ua command in SIP user-agent configuration mode.
exit sip-uaSyntax Description
This command has no arguments or keywords.
Defaults
There are no default behaviors or values for this command.
Command Modes
SIP user-agent configuration
Command History
| Release | Modification |
|---|---|
12.1(1)T | This command was introduced. |
Examples
Router (config)# sip-ua Router (config-sip-ua)# exit sip-ua
Related Commands
| Command | Description |
|---|---|
Enables the SIP user-agent configuration commands, with which you configure the user agent. |
To enable gateway specific accounting, use the gw-accounting global configuration command. To disable this function, use the no form of this command.
gw-accounting {voip | syslog | h323 [syslog] }Syntax Description
voip | This new option uses RADIUS to output accounting call data records (CDRs). Both H.323 and SIP protocols can use this method, so the name is not bound to a protocol. Use this method with the SIP feature. |
syslog | Syslog uses the system logging facility to output CDRs. |
h323 | H.323 method uses RADIUS to output accounting CDRs. |
Defaults
There are no default behaviors or values for this command.
Command Modes
Global configuration
Command History
| Release | Modification |
|---|---|
11.3(6)NA2 | This command was introduced. |
12.1(1)T | The voip option was added. |
Usage Guidelines
Examples
Router (config)# gw-accounting voip
Related Commands
| Command | Description |
|---|---|
dial-peer voice | Enters dial-peer configuration mode, and specifies the method of voice-related encapsulation. |
To specify an inband-alerting SIP header, use the inband-alerting command in SIP user-agent configuration mode. To disable this function, use the no form of this command.
inband-alerting header-stringSyntax Description
header-string | Header-string sent to SIP clients to inform them of gateway service provider (SP) behavior. |
Defaults
The default generates the header "Require: com.cisco.inband-alerting" in outgoing INVITE messages.
Command Modes
SIP user-agent configuration
Command History
| Release | Modification |
|---|---|
12.1(1)T | This command was introduced. |
Usage Guidelines
To reset this command to the default value, use the default command.
For more configuration information on inband-alerting, refer to the Session Initiation Protocol Call Flows document.
Examples
Router (config)# sip-ua Router (config-sip-ua)# inband-alerting 'Cisco inband-alerting required'
Related Commands
| Command | Description |
|---|---|
Enables the SIP user-agent configuration commands, with which you configure the user agent. |
To set the maximum number of redirects that the user agent allows, use the max-redirects command in the dial-peer configuration mode. To reset this command to the default value, use the no form of this command.
max-redirects numberSyntax Description
| number | Number of redirects: 1 through 10 are valid inputs. |
Defaults
The default number of redirects is 1.
Command Modes
Dial-peer configuration
Command History
| Release | Modification |
|---|---|
12.1(1)T | This command was introduced. |
Examples
Router (config)# dial-peer voice 102 voip Router (dial-peer-config)# max-redirects 2
Related Commands
| Command | Description |
|---|---|
dial-peer voice | Enters dial-peer configuration mode, and specifies the method of voice-related encapsulation. |
To configure the retry attempts for SIP messages, use the retry command in SIP user-agent configuration mode. To reset this command to the default value, use the no form of this command.
retry {invite number | response number | bye number | cancel number}Syntax Description
invite number | Number of INVITE retries: 1 through 10 are valid inputs; default = 6. |
response number | Number of RESPONSE retries: 1 through 10 are valid inputs; default = 6. |
bye number | Number of BYE retries: 1 through 10 are valid inputs; default = 10. |
cancel number | Number of CANCEL retries: 1 through 10 are valid inputs; default = 10. |
Defaults
Refer to the Syntax Description table for default values.
Command Modes
SIP user-agent configuration
Command History
| Release | Modification |
|---|---|
12.1(1)T | This command was introduced. |
Examples
Router (config)# sip-ua Router (config-sip-ua)# retry invite 5
Related Commands
| Command | Description |
|---|---|
Enables the sip-ua configuration commands, with which you configure the user agent. |
To configure a VoIP dial-peer to use either H323 or SIP as the session protocol for VoIP call signaling, use the session protocol command in dial-peer configuration mode. To disable this function, use the no form of this command.
session protocol {cisco | sipv2}Syntax Description
cisco | Configure the dial peer to use proprietary Cisco VoIP session protocol. |
sipv2 | SIP users should use this new option. This option configures the dial peer to use IETF SIP. |
Defaults
No default behaviors or values.
Command Modes
Dial-peer configuration
Command History
| Release | Modification |
|---|---|
11.3(1)T | This command was introduced. |
12.0(3)XG | The cisco option was added. |
12.1(1)T | The sipv2 option was added. |
Examples
Router (config)# dial-peer voice 102 voip Router (dial-peer-config)# session protocol sipv2
Related Commands
| Command | Description |
|---|---|
dial-peer voice | Enters dial-peer configuration mode, and specifies the method of voice-related encapsulation. |
To specify a network-specific address for a dial peer, use the session target command in dial-peer configuration mode. To reset this command to the default value, use the no form of this command.
session target { sip-server | dns:host-name | ipv4:ip-address[:port-number] | ras}Syntax Description
sip-server | This new option sets the session target to the global SIP server. |
dns:host-name | Indicates that the domain name server resolves the name of the IP address. A valid DNS host name is in this form: (Optional) You can use one of the following four wildcards with this keyword when defining the session target for VoIP peers:
|
ipv4:ip-address | Sets the IP address of the dial peer. A valid IP address is in this form: |
port-number | (Optional) Contact this port number to complete the call leg. |
ras | Enables the Registration, Admission, and Status (RAS) signaling function protocol so that a gatekeeper is consulted to translate the E.164 address to an IP address. |
Defaults
The default for this command is enabled with no IP address or domain name defined.
Command Modes
Dial-peer configuration
Command History
| Release | Modification |
|---|---|
11.3(1)T | This command was introduced. |
12.1(1)T | The sip-server option was added. |
Usage Guidelines
Enter the session target command to specify a network-specific address or domain name for a dial peer. Whether you select a network-specific address or a domain name depends on the session protocol you select.
You can enter the session target dns command with or without the specified wild cards. Using the optional wildcards can reduce the number of VoIP dial-peer session targets you need to configure if you have groups of numbers associated with a particular router.
Examples
Router (config)# dial-peer voice 102 voip Router (dial-peer-config)# session target dns:UA-1-f0.sip.com
Related Commands
| Command | Description |
|---|---|
dial-peer voice | Enters dial-peer configuration mode, and specifies the method of voice-related encapsulation. |
Configures the SIP server interface. |
To configure the VoIP dial peer to use TCP or UDP as the underlying transport layer protocol for SIP messages, use the session transport command in dial-peer configuration mode. To reset this command to the default value, use the no form of this command.
session transport {udp | tcp}Syntax Description
udp | Configure the SIP dial peer to use the UDP transport layer protocol. |
tcp | Configure the SIP dial peer to use the TCP transport layer protocol. |
Defaults
The default for this command is that UDP is enabled.
|
Note The transport protocol for transport and session transport must be the same. |
Command Modes
Dial-peer configuration
Command History
| Release | Modification |
|---|---|
12.1(1)T | This command was introduced. |
Usage Guidelines
Use show sip-ua status to ensure that the transport protocol that you set in session transport matches the protocol set in (config-sip-ua) # transport.
Examples
Router (config)# dial-peer voice 102 voip Router (dial-peer-config)# session transport udp
Related Commands
| Command | Description |
|---|---|
dial-peer voice | Enters dial-peer configuration mode, and specifies the method of voice-related encapsulation. |
To display statistics for SIP retires, timers, and current listener status, enter the show sip-ua command.
show sip-ua {retry | status | timers}Syntax Description
retry | Displays SIP protocol retry counts. |
status | Displays SIP UA listener status. |
timers | Displays SIP protocol timers. |
Defaults
There are no default behaviors or values for this command.
Command Modes
EXEC
Command History
| Release | Modification |
|---|---|
12.1(1)T | This command was introduced. |
Examples
The following example displays output for the show sip-ua retry command:
router#show sip-ua retry SIP UA Retry Values invite retry count = 2 response retry count = 2 bye retry count = 2 cancel retry count = 1
The following example displays output for the show sip-ua status command:
router#show sip-ua status SIP User Agent Status SIP User Agent for UDP :ENABLED SIP User Agent for TCP :ENABLED
The following example displays output for the show sip-ua timers command:
router#show sip-ua timers SIP UA Timer Values invite-wait-100 = 500 millisec invite-wait-180 = 30000 millisec invite-wait-200 = 60000 millisec 200-wait-ack = 1000 millisec bye-wait-200 = 500 millisec
Related Commands
| Command | Description |
|---|---|
Enables the SIP user-agent configuration commands, with which you configure the user agent. |
To configure the SIP server interface, use the sip-server command in SIP user-agent configuration mode. This command eliminates the need to repeatedly enter the SIP server interface in the dial peers.
sip-server {dns:host-name | ipv4:ip-address[:port-number]}Syntax Description
dns | Sets the global SIP server interface to a DNS. |
host-name | A valid DNS host name takes the following format: gateway.company.com. |
ipv4:ip-address | Sets the global SIP server interface to an IP address. A valid IP address takes the following format: xxx.xxx.xxx.xxx |
port-number | (Optional) Specifies the port number for the SIP server. |
Defaults
The default for this command is a null value.
Command Modes
SIP user-agent configuration
Command History
| Release | Modification |
|---|---|
12.1(1)T | This command was introduced. |
Examples
Router (config)# sip-ua Router (config-sip-ua)# sip-server dns:UA-1-f0.sip.com
Related Commands
| Command | Description |
|---|---|
Enables the SIP user-agent configuration commands, with which you configure the user agent. |
To enable the SIP user-agent configuration commands, with which you configure the user agent, use the sip-ua command in global configuration mode. To reset all configuration commands to their default values, use the no form of this command.
sip-uaSyntax Description
This command has no arguments or keywords.
Defaults
There are no default behaviors or values for this command.
Command Modes
Global configuration
Command History
| Release | Modification |
|---|---|
12.1(1)T | This command was introduced. |
Usage Guidelines
Enter the sip-ua command to enter the SIP user agent-configuration sub-mode. The submode configuration commands are:
| Command | Description |
Exits the SIP user agent configuration mode. | |
Specifies an inband-alerting SIP header. | |
Configures the SIP signaling timers for retry attempts. | |
Configures a SIP server interface. | |
Configures the SIP signaling timers configuration. | |
Enables or disables a SIP user agent transport for TCP or UDP, the protocol SIP user agents will be listening for on port 5060 (default). |
Examples
Router (config)# sip-ua Router (config-sip-ua)# retry invite 2 Router (config-sip-ua)# retry response 2 Router (config-sip-ua)# retry bye 2 Router (config-sip-ua)# retry cancel 2 Router (config-sip-ua)# sip-server ipv4:10.0.2.254 Router (config-sip-ua)# timers invite-wait-100 500 Router (config-sip-ua)# exit
Related Commands
| Command | Description |
|---|---|
Exits the SIP user agent configuration. | |
Specifies an inband-alerting SIP header. | |
Configures the retry attempts for SIP messages. | |
Displays statistics for SIP retires, timers, and current listener status. | |
Configures the SIP server interface. | |
Configures the SIP signaling timers. | |
Configures the SIP user agent (gateway) for SIP signaling messages on inbound calls through the SIP TCP or UDP socket. |
To configure the SIP signaling timers, use the timers command in SIP user-agent configuration mode. To reset this command to the default value, use the no form of this command.
timers {invite-wait-100 number| invite-wait-180 number | invite-wait-200 number | 200-wait-ack number | bye-wait-200 number}Syntax Description
invite-wait-100 number | Time (in milliseconds) to wait for a 100 response to an INVITE request; default = 500. |
invite-wait-180 number | Time (in milliseconds) to wait for a 180 response to an INVITE request; default = 30000. |
invite-wait-200 number | Time (in milliseconds) to wait for a 200 response to an INVITE request; default = 60000. |
200-wait-ack number | Time (in milliseconds) to wait for a 200 response to an ACK request; default = 500. |
bye-wait-200 number | Time (in milliseconds) to wait for a 200 response to a BYE request; default = 500. |
Defaults
The default is the default value for each argument as listed in the Syntax Description table.
Command Modes
SIP user-agent configuration
Command History
| Release | Modification |
|---|---|
12.1(1)T | This command was introduced. |
Examples
Router (config)# sip-ua Router (config-sip-ua)# timers invite-wait-100 500
Related Commands
| Command | Description |
|---|---|
Enables the SIP user-agent configuration commands, with which you configure the user agent. |
To configure the SIP user agent (gateway) for SIP signaling messages on inbound calls through the SIP TCP or UDP socket, use the transport command in SIP user-agent configuration mode. This command controls whether messages reach the SIP service provider interface (SPI). By setting udp or tcp as the protocol, this will be the protocol SIP user agents will be listening for on port 5060 (default). To block reception of SIP signaling messages on a specific socket, use the no form of this command.
transport {udp | tcp}Syntax Description
udp | Configures the SIP user agent to receive SIP messages on UDP port 5060. |
tcp | Configures the SIP user agent to receive SIP messages on TCP port 5060. |
Defaults
By default, sip-ua enables both UDP and TCP transport protocols.
Command Modes
SIP user-agent configuration
Command History
| Release | Modification |
|---|---|
12.1(1)T | This command was introduced. |
Examples
Router (config)# sip-ua Router (config-sip-ua)# no transport tcp
Related Commands
| Command | Description |
|---|---|
Enables the SIP user-agent configuration commands, with which you configure the user agent. |
This section documents new and modified debug commands associated with the SIP feature. All other commands used with this feature are documented in the Cisco IOS Release 12.1(1)T command references. All debug commands are EXEC commands.
To enable all SIP-related debugging, enter the debug ccsip all command. To disable all debugging output, use the no form of this command.
debug ccsip allCommand History
| Release | Modification |
|---|---|
12.1(1)T | This command was introduced. |
Usage Guidelines
The debug ccsip all command enables the following debug SIP commands:
| Command | Description |
|---|---|
Shows all SIP Service Provider Interface (SPI) call tracing. | |
Shows SIP Service Provider Interface (SPI) errors. | |
Shows all SIP Service Provider Interface (SPI) events tracing. | |
Shows all SIP Service Provider Interface (SPI) message tracing. | |
Shows all SIP Service Provider Interface (SPI) state tracing. |
Examples
UA-1#deb ccsip all All SIP call tracing enabled UA-1# *Jan 2 18:36:38:%ISDN-6-LAYER2UP:Layer 2 for Interface Se3:23, TEI 0 changed to up *Jan 2 18:36:49.302:0x621FA630 :State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE) *Jan 2 18:36:49.302: Queued event from SIP SPI :SIPSPI_EV_CC_CALL_SETUP *Jan 2 18:36:49.302:CCSIP-SPI-CONTROL: act_idle_call_setup *Jan 2 18:36:49.302: act_idle_call_setup:Not using Voice Class Codec *Jan 2 18:36:49.302:act_idle_call_setup:preferred_codec set[0] type :g711ulaw bytes:160 *Jan 2 18:36:49.302: Queued event from SIP SPI :SIPSPI_EV_CREATE_CONNECTION *Jan 2 18:36:49.306:0x621FA630 :State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_CONNECTING) *Jan 2 18:36:49.306:0x621FA630 :State change from (STATE_IDLE, SUBSTATE_CONNECTING) to (STATE_IDLE, SUBSTATE_CONNECTING) *Jan 2 18:36:49.306:CCSIP-SPI-CONTROL: act_idle_connection_created *Jan 2 18:36:49.306:CCSIP-SPI-CONTROL: act_idle_connection_created:Connid(1) created to 2.0.0.2:5060, local_port 6932 *Jan 2 18:36:49.310: Queued event from SIP SPI :SIPSPI_EV_SEND_MESSAGE *Jan 2 18:36:49.310:0x621FA630 :State change from (STATE_IDLE, SUBSTATE_CONNECTING) to (STATE_SENT_INVITE, SUBSTATE_NONE) *Jan 2 18:36:49.310: Send: INVITE sip:9605550001@10.0.0.2;user=phone SIP/2.0 Via:SIP/2.0/UDP 10.0.0.1:6932 From:sip:9505550001@10.0.0.1 To:<sip:9605550001@10.0.0.2;user=phone> Date:Sun 02 Jan 2000 14:36:49 EDT Call-ID:CA954057-B701C020-0-924EB2C@10.0.2.2 Cisco-Guid:3398778967-3070345246-0-153414444 Require:com.cisco.inband-alerting User-Agent:Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq:100 INVITE Content-Type:application/sdp Content-Length:113 v=0 o=CiscoSystemsSIPUserAgent 6659 1152 IN IP4 10.0.0.1 s=SIP Call c=IN IP4 10.0.0.1 m=audio 20910 RTP/AVP 0 *Jan 2 18:36:49.318:Received : SIP/2.0 100 Trying Via:SIP/2.0/UDP 10.0.0.1:6932 From:sip:9505550001@10.0.0.1 To:<sip:9605550001@10.0.0.2;user=phone> Date:Sun 02 Jan 2000 14:36:46 EDT Call-ID:CA954057-B701C020-0-924EB2C@10.0.2.2 Server:Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq:100 INVITE Content-Length:0 *Jan 2 18:36:49.318:HandleUdpSocketReads :Msg enqueued for SPI with IPaddr:10.0.0.2:5060 *Jan 2 18:36:49.318:CCSIP-SPI-CONTROL: act_sentinvite_new_message *Jan 2 18:36:49.318:CCSIP-SPI-CONTROL: sipSPICheckResponse *Jan 2 18:36:49.318:0x621FA630 :State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING) *Jan 2 18:36:49.362:Received: SIP/2.0 180 Ringing Via:SIP/2.0/UDP 10.0.0.1:6932 From:sip:9505550001@10.0.0.1 To:<sip:9605550001@10.0.0.2;user=phone> Date:Sun 02 Jan 2000 14:36:46 EDT Call-ID:CA954057-B701C020-0-924EB2C@10.0.2.2 Server:Cisco VoIP Gateway/ IOS 12.x/ SIP enabled Content-Type:application/sdp CSeq:100 INVITE Content-Length:113 v=0 o=CiscoSystemsSIPUserAgent 7548 9300 IN IP4 10.0.0.2 s=SIP Call c=IN IP4 10.0.0.2 m=audio 20234 RTP/AVP 0 *Jan 2 18:36:49.362:HandleUdpSocketReads:Msg enqueued for SPI with IPaddr:10.0.0.2:5060 *Jan 2 18:36:49.362:CCSIP-SPI-CONTROL: act_recdproc_new_message *Jan 2 18:36:49.362:CCSIP-SPI-CONTROL: sipSPICheckResponse *Jan 2 18:36:49.362:CCSIP-SPI-CONTROL: sipSPICheckResponse:Updating session description *Jan 2 18:36:49.362:CCSIP-SPI-CONTROL: act_recdproc_new_message:SDP MediaTypes negotiation successful! Negotiated Codec :g711ulaw , bytes :160 Inband Alerting :2 *Jan 2 18:36:49.366:0x621FA630 :State change from (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_ALERTING) *Jan 2 18:36:49.366:CCSIP-SPI-CONTROL: ccsip_caps_ind *Jan 2 18:36:49.366:ccsip_caps_ind:Load DSP with codec (5) g711ulaw, Bytes=160 *Jan 2 18:36:49.366:ccsip_caps_ind:set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_INBAND_VOICE *Jan 2 18:36:49.366:CCSIP-SPI-CONTROL: ccsip_caps_ack *Jan 2 18:36:49.782:Received: SIP/2.0 200 OK Via:SIP/2.0/UDP 10.0.0.1:6932 From:sip:9505550001@10.0.0.1 To:<sip:9605550001@10.0.0.2;user=phone> Date:Sun 02 Jan 2000 14:36:46 EDT Call-ID:CA954057-B701C020-0-924EB2C@10.0.2.2 Server:Cisco VoIP Gateway/ IOS 12.x/ SIP enabled Content-Type:application/sdp CSeq:100 INVITE Content-Length:113 v=0 o=CiscoSystemsSIPUserAgent 6822 5961 IN IP4 10.0.0.2 s=SIP Call c=IN IP4 10.0.0.2 m=audio 20234 RTP/AVP 0 *Jan 2 18:36:49.786:HandleUdpSocketReads :Msg enqueued for SPI with IPaddr:10.0.0.2:5060 *Jan 2 18:36:49.786:CCSIP-SPI-CONTROL: act_recdproc_new_message *Jan 2 18:36:49.786:CCSIP-SPI-CONTROL: sipSPICheckResponse *Jan 2 18:36:49.786:CCSIP-SPI-CONTROL: sipSPICheckResponse:Updating session description *Jan 2 18:36:49.786:CCSIP-SPI-CONTROL: act_recdproc_new_message:SDP MediaTypes negotiation successful! Negotiated Codec :g711ulaw, bytes:160 *Jan 2 18:36:49.786: Queued event from SIP SPI :SIPSPI_EV_SEND_MESSAGE *Jan 2 18:36:49.786:0x621FA630 :State change from (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_ALERTING) to (STATE_ACTIVE, SUBSTATE_NONE) *Jan 2 18:36:49.786:The Call Setup Information is: Call Control Block (CCB) :0x621FA630 State of The Call :STATE_ACTIVE TCP Sockets Used :NO Calling Number :9505550001 Called Number :9605550001 Negotiated Codec :g711ulaw Source IP Address (Media):10.0.0.1 Source IP Port (Media):20910 Destn IP Address (Media):10.0.0.2 Destn IP Port (Media):20234 Destn SIP Addr (Control) :10.0.0.2 Destn SIP Port (Control) :5060 Destination Name :10.0.0.2 *Jan 2 18:36:49.790: Send: ACK sip:9605550001@10.0.0.2;user=phone SIP/2.0 Via:SIP/2.0/UDP 10.0.0.1:6932 From:sip:9505550001@10.0.0.1 To:<sip:9605550001@10.0.0.2;user=phone> Date:Sun 02 Jan 2000 14:36:49 EDT Call-ID:CA954057-B701C020-0-924EB2C@10.0.2.2 Content-Type:application/sdp Content-Length:113 CSeq:100 ACK v=0 o=CiscoSystemsSIPUserAgent 8267 3722 IN IP4 10.0.0.1 s=SIP Call c=IN IP4 10.0.0.1 m=audio 20910 RTP/AVP 0 *Jan 2 18:37:20.893: Queued event From SIP SPI to CCAPI/DNS:SIPSPI_EV_CC_CALL_DISCONNECT *Jan 2 18:37:20.893:CCSIP-SPI-CONTROL: act_active_disconnect *Jan 2 18:37:20.893: Queued event from SIP SPI :SIPSPI_EV_SEND_MESSAGE *Jan 2 18:37:20.893:0x621FA630 :State change from (STATE_ACTIVE, SUBSTATE_NONE) to (STATE_DISCONNECTING, SUBSTATE_NONE) *Jan 2 18:37:20.897: Send: BYE <sip:9605550001@10.0.0.2;user=phone> SIP/2.0 Via:SIP/2.0/UDP 10.0.0.1:6932 From:sip:9505550001@10.0.0.1 To:<sip:9605550001@10.0.0.2;user=phone> Date:Sun 02 Jan 2000 14:36:49 EDT Call-ID:CA954057-B701C020-0-924EB2C@10.0.2.2 User-Agent:Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq:101 BYE Content-Length:0 *Jan 2 18:37:20.901:Received: SIP/2.0 200 OK Via:SIP/2.0/UDP 10.0.0.1:6932 From:sip:9505550001@10.0.0.1 To:<sip:9605550001@10.0.0.2;user=phone> Date:Sun 02 Jan 2000 14:37:18 EDT Call-ID:CA954057-B701C020-0-924EB2C@10.0.2.2 Server:Cisco VoIP Gateway/ IOS 12.x/ SIP enabled Content-Length:0 CSeq:101 BYE *Jan 2 18:37:20.901:HandleUdpSocketReads :Msg enqueued for SPI with IPaddr:10.0.0.2:5060 *Jan 2 18:37:20.901:CCSIP-SPI-CONTROL: act_disconnecting_new_message *Jan 2 18:37:20.901:CCSIP-SPI-CONTROL: sact_disconnecting_new_message_response *Jan 2 18:37:20.901:CCSIP-SPI-CONTROL: sipSPICheckResponse *Jan 2 18:37:20.901:CCSIP-SPI-CONTROL: sipSPICallCleanup *Jan 2 18:37:20.901: Queued event from SIP SPI :SIPSPI_EV_CLOSE_CONNECTION *Jan 2 18:37:20.905:0x621FA630 :State change from (STATE_DISCONNECTING, SUBSTATE_NONE) to (STATE_DEAD, SUBSTATE_NONE) *Jan 2 18:37:20.905:The Call Setup Information is : Call Control Block (CCB) :0x621FA630 State of The Call :STATE_DEAD TCP Sockets Used :NO Calling Number :9505550001 Called Number :9605550001 Negotiated Codec :g711ulaw Source IP Address (Media):10.0.0.1 Source IP Port (Media):20910 Destn IP Address (Media):10.0.0.2 Destn IP Port (Media):20234 Destn SIP Addr (Control) :10.0.0.2 Destn SIP Port (Control) :5060 Destination Name :10.0.0.2 *Jan 2 18:37:20.905: Disconnect Cause (CC) :16 Disconnect Cause (SIP) :200
To show all SIP Service Provider Interface (SPI) call tracing, enter the debug ccsip calls command. This command traces the SIP call details as updated in the SIP call control block.
debug ccsip callsCommand History
| Release | Modification |
|---|---|
12.1(1)T | This command was introduced. |
Examples
UA-1#deb ccsip calls SIP Call statistics tracing is enabled UA-1# *Jan 2 18:21:01.245: The Call Setup Information is: Call Control Block (CCB): 0x62202E3C State of The Call : STATE_ACTIVE TCP Sockets Used : NO Calling Number : 9505550001 Called Number : 9605550001 Negotiated Codec : g711ulaw Source IP Address (Media): 10.0.0.1 Source IP Port (Media): 20754 Destn IP Address (Media): 10.0.0.2 Destn IP Port (Media): 20748 Destn SIP Addr (Control): 10.0.0.2 Destn SIP Port (Control): 5060 Destination Name : 10.0.0.2 *Jan 2 18:21:32.708: The Call Setup Information is : Call Control Block (CCB): 0x62202E3C State of The Call : STATE_DEAD TCP Sockets Used : NO Calling Number : 9505550001 Called Number : 9605550001 Negotiated Codec : g711ulaw Source IP Address (Media): 10.0.0.1 Source IP Port (Media): 20754 Destn IP Address (Media): 10.0.0.2 Destn IP Port (Media): 20748 Destn SIP Addr (Control): 10.0.0.2 Destn SIP Port (Control): 5060 Destination Name : 10.0.0.2 *Jan 2 18:21:32.708: Disconnect Cause (CC) : 16 Disconnect Cause (SIP) : 200
To show SIP Service Provider Interface (SPI) errors, enter the debug ccsip error command. This command traces all error messages generated from errors encountered by the SIP subsystem.
debug ccsip errorCommand History
| Release | Modification |
|---|---|
12.1(1)T | This command was introduced. |
Examples
UA-1#deb ccsip error SIP Call error tracing is enabled UA-1# *Jan 2 18:24:25.281: CCSIP-SPI-CONTROL: act_idle_call_setup *Jan 2 18:24:25.281: act_idle_call_setup:Not using Voice Class Codec *Jan 2 18:24:25.281: act_idle_call_setup: preferred_codec set[0] type :g711ulaw bytes: 160 *Jan 2 18:24:25.281: CCSIP-SPI-CONTROL: act_idle_connection_created *Jan 2 18:24:25.285: CCSIP-SPI-CONTROL: act_idle_connection_created: Connid(1) created to 2.0.0.2:5060, local_port 9830 *Jan 2 18:24:25.293: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 2.0.0.2:5060 *Jan 2 18:24:25.293: CCSIP-SPI-CONTROL: act_sentinvite_new_message *Jan 2 18:24:25.293: CCSIP-SPI-CONTROL: sipSPICheckResponse *Jan 2 18:24:25.337: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 2.0.0.2:5060 *Jan 2 18:24:25.341: CCSIP-SPI-CONTROL: act_recdproc_new_message *Jan 2 18:24:25.341: CCSIP-SPI-CONTROL: sipSPICheckResponse *Jan 2 18:24:25.341: CCSIP-SPI-CONTROL: sipSPICheckResponse : Updating session description *Jan 2 18:24:25.341: CCSIP-SPI-CONTROL: act_recdproc_new_message: SDP MediaTypes negotiation successful! Negotiated Codec : g711ulaw , bytes :160 Inband Alerting : 2 *Jan 2 18:24:25.341: CCSIP-SPI-CONTROL: ccsip_caps_ind *Jan 2 18:24:25.341: ccsip_caps_ind: Load DSP with codec (5) g711ulaw, Bytes=160 *Jan 2 18:24:25.341: ccsip_caps_ind: set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_INBAND_VOICE *Jan 2 18:24:25.341: CCSIP-SPI-CONTROL: ccsip_caps_ack *Jan 2 18:24:25.769: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 2.0.0.2:5060 *Jan 2 18:24:25.773: CCSIP-SPI-CONTROL: act_recdproc_new_message *Jan 2 18:24:25.773: CCSIP-SPI-CONTROL: sipSPICheckResponse *Jan 2 18:24:25.773: CCSIP-SPI-CONTROL: sipSPICheckResponse : Updating session description *Jan 2 18:24:25.773: CCSIP-SPI-CONTROL: act_recdproc_new_message: SDP MediaTypes negotiation successful! Negotiated Codec : g711ulaw , bytes :160 *Jan 2 18:24:57.012: CCSIP-SPI-CONTROL: act_active_disconnect *Jan 2 18:24:57.020: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 2.0.0.2:5060 *Jan 2 18:24:57.020: CCSIP-SPI-CONTROL: act_disconnecting_new_message *Jan 2 18:24:57.020: CCSIP-SPI-CONTROL: sact_disconnecting_new_message_response *Jan 2 18:24:57.020: CCSIP-SPI-CONTROL: sipSPICheckResponse *Jan 2 18:24:57.020: CCSIP-SPI-CONTROL: sipSPICallCleanup
To show all SIP Service Provider Interface (SPI) events tracing, enter the debug ccsip events command. This command traces the events posted to SIP SPI from all interfaces.
debug ccsip eventsCommand History
| Release | Modification |
|---|---|
12.1(1)T | This command was introduced. |
Examples
UA-1#debug ccsip events SIP Call events tracing is enabled UA-1# *Jan 2 18:28:06.784: Queued event from SIP SPI :SIPSPI_EV_CC_CALL_SETUP *Jan 2 18:28:06.784: Queued event from SIP SPI :SIPSPI_EV_CREATE_CONNECTION *Jan 2 18:28:06.792: Queued event from SIP SPI :SIPSPI_EV_SEND_MESSAGE *Jan 2 18:28:07.284: Queued event from SIP SPI :SIPSPI_EV_SEND_MESSAGE *Jan 2 18:28:38.384: Queued event From SIP SPI to CCAPI/DNS:SIPSPI_EV_CC_CALL_DISCONNECT *Jan 2 18:28:38.388: Queued event from SIP SPI :SIPSPI_EV_SEND_MESSAGE
To show all SIP Service Provider Interface (SPI) message tracing, enter the debug ccsip messages command. This command traces the SIP messages exchanged between the SIP user agent client (UAC) and the access server.
debug ccsip messagesCommand History
| Release | Modification |
|---|---|
12.1(1)T | This command was introduced. |
Examples
UA-1#deb ccsip message SIP Call messages tracing is enabled UA-1# *Jan 2 20:40:40.937: Send: INVITE sip:9605550001@10.0.0.2;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.0.1:2537 From: "SIP Gateway UA-1; T1-pri controller 2" <sip:9505550001@10.0.0.1> To: <sip:9605550001@10.0.0.2;user=phone> Date: Sun 02 Jan 2000 16:40:40 EDT Call-ID: CA954057-B701C03B-0-996518C@10.0.2.2 Cisco-Guid: 3398778967-3070345273-0-160846216 Require: com.cisco.inband-alerting User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 100 INVITE Content-Type: application/sdp Content-Length: 113 v=0 o=CiscoSystemsSIPUserAgent 9074 8380 IN IP4 10.0.0.1 s=SIP Call c=IN IP4 10.0.0.1 m=audio 20610 RTP/AVP 0 *Jan 2 20:40:40.945: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.1:2537 From: "SIP Gateway UA-1; T1-pri controller 2" <sip:9505550001@10.0.0.1> To: <sip:9605550001@10.0.0.2;user=phone> Date: Sun 02 Jan 2000 16:40:38 EDT Call-ID: CA954057-B701C03B-0-996518C@10.0.2.2 Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 100 INVITE Content-Length: 0 *Jan 2 20:40:40.993: Received: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.1:2537 From: "SIP Gateway UA-1; T1-pri controller 2" <sip:9505550001@10.0.0.1> To: <sip:9605550001@10.0.0.2;user=phone> Date: Sun 02 Jan 2000 16:40:38 EDT Call-ID: CA954057-B701C03B-0-996518C@10.0.2.2 Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled Content-Type: application/sdp CSeq: 100 INVITE Content-Length: 113 v=0 o=CiscoSystemsSIPUserAgent 6706 3098 IN IP4 10.0.0.2 s=SIP Call c=IN IP4 10.0.0.2 m=audio 20460 RTP/AVP 0 *Jan 2 20:40:41.421: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.1:2537 From: "SIP Gateway UA-1; T1-pri controller 2" <sip:9505550001@10.0.0.1> To: <sip:9605550001@10.0.0.2;user=phone> Date: Sun 02 Jan 2000 16:40:38 EDT Call-ID: CA954057-B701C03B-0-996518C@10.0.2.2 Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled Content-Type: application/sdp CSeq: 100 INVITE Content-Length: 113 v=0 o=CiscoSystemsSIPUserAgent 7641 2845 IN IP4 10.0.0.2 s=SIP Call c=IN IP4 10.0.0.2 m=audio 20460 RTP/AVP 0 *Jan 2 20:40:41.425: Send: ACK sip:9605550001@10.0.0.2;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.0.1:2537 From: "SIP Gateway UA-1; T1-pri controller 2" <sip:9505550001@10.0.0.1> To: <sip:9605550001@10.0.0.2;user=phone> Date: Sun 02 Jan 2000 16:40:40 EDT Call-ID: CA954057-B701C03B-0-996518C@10.0.2.2 Content-Type: application/sdp Content-Length: 113 CSeq: 100 ACK v=0 o=CiscoSystemsSIPUserAgent 6286 8863 IN IP4 10.0.0.1 s=SIP Call c=IN IP4 10.0.0.1 m=audio 20610 RTP/AVP 0 *Jan 2 20:41:12.596: Send: BYE <sip:9605550001@10.0.0.2;user=phone> SIP/2.0 Via: SIP/2.0/UDP 10.0.0.1:2537 From: "SIP Gateway UA-1; T1-pri controller 2" <sip:9505550001@10.0.0.1> To: <sip:9605550001@10.0.0.2;user=phone> Date: Sun 02 Jan 2000 16:40:40 EDT Call-ID: CA954057-B701C03B-0-996518C@10.0.2.2 User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 101 BYE Content-Length: 0 *Jan 2 20:41:12.600: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.1:2537 From: "SIP Gateway UA-1; T1-pri controller 2" <sip:9505550001@10.0.0.1> To: <sip:9605550001@10.0.0.2;user=phone> Date: Sun 02 Jan 2000 16:41:09 EDT Call-ID: CA954057-B701C03B-0-996518C@10.0.2.2 Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled Content-Length: 0 CSeq: 101 BYE
To show all SIP Service Provider Interface (SPI) state tracing, enter the debug ccsip states command. This command traces the state machine changes of SIP SPI and displays the state transitions.
debug ccsip statesCommand History
| Release | Modification |
|---|---|
12.1(1)T | This command was introduced. |
Examples
UA-1#deb ccsip states SIP Call states tracing is enabled UA-1# *Jan 2 18:34:37.793:0x6220C634 :State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE) *Jan 2 18:34:37.797:0x6220C634 :State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_CONNECTING) *Jan 2 18:34:37.797:0x6220C634 :State change from (STATE_IDLE, SUBSTATE_CONNECTING) to (STATE_IDLE, SUBSTATE_CONNECTING) *Jan 2 18:34:37.801:0x6220C634 :State change from (STATE_IDLE, SUBSTATE_CONNECTING) to (STATE_SENT_INVITE, SUBSTATE_NONE) *Jan 2 18:34:37.809:0x6220C634 :State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING) *Jan 2 18:34:37.853:0x6220C634 :State change from (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_ALERTING) *Jan 2 18:34:38.261:0x6220C634 :State change from (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_ALERTING) to (STATE_ACTIVE, SUBSTATE_NONE) *Jan 2 18:35:09.860:0x6220C634 :State change from (STATE_ACTIVE, SUBSTATE_NONE) to (STATE_DISCONNECTING, SUBSTATE_NONE) *Jan 2 18:35:09.868:0x6220C634 :State change from (STATE_DISCONNECTING, SUBSTATE_NONE) to (STATE_DEAD, SUBSTATE_NONE) *Jan 2 18:28:38.404: Queued event from SIP SPI :SIPSPI_EV_CLOSE_CONNECTION
AAA---authentication, authorization, and accounting. AAA is a suite of network security services that provides the primary framework through which access control can be set up on your Cisco router or access server.
ANI---automatic number identification. SS7 (signaling system 7) feature in which a series of digits, either analog or digital, are included in the call, identifying the telephone number of the calling device. In other words, ANI identifies the number of the calling party.
CAS---channel associated signaling.
CCAPI---call control applications programming interface.
CLI---command line interface. Interface that allows the user to interact with the operating system by entering commands and optional arguments. The UNIX operating system and DOS provide CLIs.
CO---central office. Local telephone company office to which all local loops in a given area connect and in which circuit switching of subscriber lines occurs.
CPE---customer premises equipment. Terminating equipment, such as terminals, telephones, and modems, supplied by the telephone company, installed at the customer sites, and connected to the telephone company network.
CSM---call switching module.
dial peer---An addressable call endpoint. In Voice over IP (VoIP), there are two types of dial peers: POTS and VoIP.
DNS---Domain Name System. System used in the Internet for translating names of network nodes into addresses.
DNIS---dialed number identification service (the called number).
DSP---digital signal processor.
DTMF---dual tone multifrequency. Use of two simultaneous voice-band tones for dialing (such as touch tone).
E.164---The international public telecommunications numbering plan. A standard set by ITU-T which addresses telephone numbers.
E&M---recEive and transMit (or ear and mouth). Trunking arrangement generally used for two-way switch-to-switch or switch-to-network connections. Cisco's analog E&M interface is an RJ-48 connector that allows connections to PBX trunk lines (tie lines). E&M is also available on E1 and T1 digital interfaces.
endpoint---A H.323 terminal or gateway. An endpoint can call and be called. It generates and/or terminates the information stream.
gateway---A gateway allows SIP or H.323 terminals to communicate with terminals configured to other protocols by converting protocols. A gateway is the point where a circuit-switched call is encoded and repackaged into IP packets.
H.323---An International Telecommunication Union (ITU-T) standard that describes packet-based video, audio, and data conferencing. H.323 is an umbrella standard that describes the architecture of the conferencing system and refers to a set of other standards (H.245, H.225.0, and Q.931) to describe its actual protocol.
H.323 RAS---registration, admission, and status. The RAS signaling function performs registration, admissions, bandwidth changes, status and disengage procedures between the VoIP gateway and the gatekeeper.
IVR---Integrated voice response. When someone dials in, IVR responds with a prompt to get a personal identification number (PIN), and so on.
LEC---local exchange carrier. Local or regional telephone company that owns and operates a telephone network and the customer lines that connect to it.
Location Server---A SIP redirect or proxy server uses a a location service to get information about a caller's location(s). Location services are offered by location servers.
MF---Multifrequency tones are made of six frequencies that provide 15 two frequency combinations for indication digits 0-9 and KP/ST signals.
multicast---A process of transmitting PDUs from one source to many destinations. The actual mechanism (that is, IP multicast, multi-unicast, and so forth) for this process might be different for LAN technologies.
multipoint-unicast---A process of transferring PDUs (Protocol Data Units) where an endpoint sends more than one copy of a media stream to different endpoints. This can be necessary in networks which do not support multicast.
node---An H.323 entity that uses RAS to communicate with the gatekeeper, for example, an endpoint such as a terminal, proxy, or gateway.
PDU---Protocol data units used by bridges to transfer connectivity information.
POTS---Plain old telephone service. Basic telephone service supplying standard single line telephones, telephone lines, and access to the PSTN.
Proxy Server---An intermediary program that acts as both a server and a client for the purpose of making requests on behalf of other clients. Requests are serviced internally or by passing them on, possibly after translation, to other servers. A proxy interprets, and, if necessary, rewrites a request message before forwarding it.
Redirect Server---A redirect server is a server that accepts a SIP request, maps the address into zero or more new addresses and returns these addresses to the client. It does not initiate its own SIP request nor accept calls.
Registrar---A registrar is a server that accepts REGISTER requests. A registrar is typically co-located with a proxy or redirect server and MAY offer location services.
PSTN---Public switched telephone network. PSTN refers to the local telephone company.
RAS---Registration, admission, and status protocol. This is the protocol that is used between endpoints and the gatekeeper to perform management functions.
RBS---robbed bit signaling.
SIP---Session Initiation Protocol. This is a protocol developed by the IETF MMUSIC Working Group as an alternative to H.323. SIP features are compliant with IETF RFC 2543, published in March 1999.
SIP equips platforms to signal the setup of voice and multimedia calls over IP networks.
SPI---service provider interface.
TDM---Time-division multiplexing. Technique in which information from multiple channels can be allocated bandwidth on a single wire based on preassigned time slots. Bandwidth is allocated to each channel regardless of whether the station has data to transmit.
User Agent---see UAS.
UAC---User Agent Client: A user agent client is a client application that initiates the SIP request.
UAS---User Agent Server (or user agent): A user agent server is a server application that contacts the user when a SIP request is received, then returns a response on behalf of the user. The response accepts, rejects or redirects the request.
VoIP---Voice over IP. The ability to carry normal telephone-style voice over an IP-based Internet with POTs-like functionality, reliability, and voice quality. VoIP is a blanket term, which generally refers to Cisco's standards based (for example H.323) approach to IP voice traffic.
Posted: Mon Jul 31 14:53:51 PDT 2000
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