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Voice over Internet Protocol (VoIP) currently implements the ITU's H.323 specification within Internet Telephony Gateways (ITGs) to signal voice call setup. The Session Initiation Protocol (SIP) is a new protocol developed by the Internet Engineering Task Force (IETF) for multimedia conferencing over IP. SIP features are compliant with IETF RFC 2543, SIP: Session Initiation Protocol, published in March 1999.
The Cisco SIP functionality, introduced in Cisco IOS Release 12.1(1)T and enhanced in Cisco IOS Release 12.1(3)T, enables Cisco access platforms to signal the setup of voice and multimedia calls over IP networks. The SIP feature also provides non-proprietary advantages in the areas of:
This document includes the following sections:
The SIP feature enhancements include the following:
| Codec | SDP |
|---|---|
G711ulaw | 0 |
G711alaw | 8 |
G723r63 | 4 |
G726r16 | 2 |
G728 | 15 |
G729r8 | 18 |
The SIP feature enhancements enable SIP gateways to:
This section lists the restrictions and considerations of the Enhancements to SIP for VoIP on Cisco Access Platforms feature:
The SIP feature is dependent upon the interoperability of Service Provider Features for VoIP.
The following documents contain information related to the Cisco SIP functionality:
Standards
No new or modified standards are supported by this feature.
MIBs
No new or modified MIBs are supported by this feature.
For descriptions of supported MIBs and how to use MIBs, see the Cisco MIB web site on CC O.
RFCs
RFC 2543 is supported by this feature.
Platform | Required DRAM | Required Flash | DSP/VCWARE |
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To configure SIP on the Cisco AS5300 access server, the Cisco 2600, or the Cisco 3600 series router, perform the tasks described in the following sections:
To configure SIP support for a VoIP dial peer, you must enter the following commands beginning in global configuration mode:
| Step | Command | Purpose |
|---|---|---|
1. | Router (config) | Enter the dial-peer mode to configure a VoIP dial peer. |
2. | Router (config-dial-peer) # session transport {udp | tcp}
| Enter the session transport type for the SIP user agent. |
3. | Router (config-dial-peer) # session protocol sipv2 | Enter the session protocol type. |
4. | Router (config-dial-peer) # | Specify the dial peer session target to use the global SIP server. |
It is not necessary to configure a SIP UA in order to place a call. A SIP UA is configured to listen by default. However, if you want to adjust any of the settings you can do so using the following commands:
| Step | Command | Purpose |
|---|---|---|
1. | Router (config) # sip-ua | Enter the SIP user agent (sip-ua) mode to configure SIP-UA related commands. |
2. | Router (config-sip-ua)# transport { udp | tcp }
| Configure the SIP user agent (sip-ua) for SIP signaling messages. The default value is udp. |
3. | Router (config-sip-ua) | Enter the host name or IP address of the SIP server interface. |
4. | Router (config-sip-ua) | Set time to wait for a response. |
5. | Router (config-sip-ua)# timers expires time | Limit the time duration (in milliseconds) of a search for an INVITE. |
6. | Router (config-sip-ua)# retry invite number | Configure the SIP signaling timers for retry attempts. |
7. | Router (config-sip-ua)# max-forwards number_of_hops | Limit the number of proxy or redirect servers that can forward a request. |
To configure SIP call transfer for a POTS dial peer, enter the following commands beginning in global configuration mode:
| Step | Command | Purpose |
|---|---|---|
1. | Router (config) | Enter the dial-peer mode to configure a POTS dial peer. |
2. | Router (config-dial-peer) # application session | Specifies that the standard session application will be invoked for this dial peer. |
3. | Router (config-dial-peer) # destination-pattern pattern | Specifies the telephone number associated with the dial peer. |
4. | Router (config-dial-peer) # | Specifies the voice slot number and port through which incoming VoIP calls will be received. |
To configure SIP call transfer for a VoIP dial peer, enter the following commands beginning in global configuration mode:
| Step | Command | Purpose |
|---|---|---|
1. | Router (config) | Enter the dial-peer mode to configure a VoIP dial peer. |
2. | Router (config-dial-peer) # application session | Specifies that the standard session application will be invoked for this dial peer. |
3. | Router (config-dial-peer) # destination-pattern pattern | Specifies the telephone number associated with the dial peer. |
4. | Router (config-dial-peer) # | Specifies the IP address of the destination gateway for outbound dial peers. |
By default, the SIP gateway tags called numbers that have 11 or more digits as "international" when sending SETUP messages to the PSTN switch. In some cases, such as situations where the user must dial 9 to access an outside line, this assumption may not be correct.
To accommodate such situations, you can define translation rules on the outbound POTS dial peer to convert the "type of number" to the correct value. Translation rules manipulate the called number digits and the "type of number" value associated with the called digits.
To define translation rules on a POTS dial peer, enter the following commands beginning in global configuration mode:
| Step | Command | Purpose |
|---|---|---|
1. | Router (config) | Defines a translation-rule tag number and enters translation-rule configuration mode. All subsequent commands that you enter in this mode before you exit will apply to this translation-rule tag. |
2. | Router (config-translate) # Rule precedence input_searched_pattern substituted_pattern [[match-type] [substituted-type]] | Specifies translation rules. This command can be entered multiple times and is applied to the translation-rule defined in Step 1. |
3. | Router (config) | Enter the dial-peer mode to configure a POTS dial peer. |
4. | Router (config-dial-peer) # translate-outgoing called name-tag | Specifies the translation tag for an outbound called number. |
5. | Router(config-dial-peer)# port slot-number/port | Specifies the voice port. |
For more information about the commands used to configure translation rules, see the Dial Peer Enhancements documentation on CCO.
The following example illustrates a translation rule for dialing national numbers in the situation where the user must dial 9 to access an outside line. In the rule command in this example:
The result of this command is that any outgoing call that is destined for a number that starts with 91 and that is considered by the gateway to be an international number, will be sent to the PSTN as a national number with a prefix of 1.
translation-rule 10 Rule 1 91% 1 international national ! ! ! dial-peer voice 10 pots destination-pattern 91.......... translate-outgoing called 10 port 1:D !
The following example illustrates a translation rule for dialing national numbers in the situation where the user does not need to dial 9 to access an outside line.
translation-rule 10 Rule 1 1% 1 international national ! ! ! dial-peer voice 10 pots destination-pattern 1.......... translate-outgoing called 10 port 1:D prefix 1 !
The following example illustrates a translation rule for dialing international numbers in the situation where the user must dial 9 to access an outside line.
translation-rule 20 Rule 1 9011% 011 unknown international ! ! ! dial-peer voice 10 pots destination-pattern 9011T translate-outgoing called 20 port 1:D !
The following example illustrates a translation rule for dialing international numbers in the situation where the user does not need to dial 9 to access an outside line.
translation-rule 20 Rule 1 011% 011 unknown international ! ! ! dial-peer voice 10 pots destination-pattern 011T translate-outgoing called 20 port 1:D prefix 011 !
Enter the show running configuration command to verify your configuration.
This section contains examples of the following:
The following shows an example of the output that appears when you enter the show running configuration command.
router1#show running config Building configuration... Current configuration: ! version 12.1 service timestamps debug datetime service timestamps log uptime no service password-encryption ! hostname router1 ! ! ! clock timezone GMT 5 voice-card 1 ! ip subnet-zero ip tcp path-mtu-discovery ip name-server 172.18.192.48 ! isdn voice-call-failure 0 ! ! controller T1 1/0 framing esf clock source line primary linecode b8zs ! controller T1 1/1 ! ! voice-port 2/0/0 ! voice-port 2/0/1 ! voice class codec 1 codec preference 1 g711alaw codec preference 2 g723r63 codec preference 3 g723r53 ! ! dial-peer voice 100 pots destination-pattern 3660110 port 2/0/0 ! dial-peer voice 200 pots application session destination-pattern 3660120 port 2/0/1 ! dial-peer voice 101 voip destination-pattern 3660210 session protocol sipv2 session target ipv4:166.34.244.73 codec g711ulaw ! dial-peer voice 201 voip application sesion destination-pattern 3660220 session protocol sipv2 session target dns:3660-2.sip.com codec g711ulaw ! dial-peer voice 999 voip destination-pattern 5551111 session protocol sipv2 session target ipv4:161.44.53.89 session transport tcp ! dial-peer voice 300 pots destination-pattern 2101100 ! dial-peer voice 350 voip destination-pattern 3100607 session protocol sipv2 session target ipv4:172.18.192.197 codec g711ulaw ! dial-peer voice 301 voip application session destination-pattern 1234 session protocol sipv2 session target ipv4:172.18.192.193 codec g711ulaw ! dial-peer voice 333 voip application session destination-pattern 1235 session protocol sipv2 session target ipv4:172.18.192.199 codec g711ulaw ! dial-peer voice 888 voip destination-pattern 888 session protocol sipv2 session target ipv4:161.44.53.89 session transport tcp codec g711ulaw ! dial-peer voice 260011 voip destination-pattern 260011 session protocol sipv2 session target ipv4:172.18.192.164 codec g711ulaw ! dial-peer voice 444 voip destination-pattern 2339000 session protocol sipv2 session target ipv4:172.18.192.205 codec g711ulaw ! dial-peer voice 111 voip destination-pattern 111 session protocol sipv2 session target sip-server codec g711ulaw ! dial-peer voice 7777777 voip destination-pattern 19197777777 session protocol sipv2 session target ipv4:172.18.192.38 codec g711ulaw ! ! sip-ua retry invite 2 retry response 2 retry bye 2 retry cancel 2 no inband-alerting sip-server dns: ! ! interface FastEthernet0/0 ip address 172.18.192.194 255.255.255.0 load-interval 30 speed auto half-duplex ! interface FastEthernet0/1 ip address 166.34.245.230 255.255.255.224 load-interval 30 speed auto half-duplex ! ip classless ip route 0.0.0.0 0.0.0.0 172.18.192.1 ip route 166.34.0.0 255.255.0.0 166.34.245.225 no ip http server ! access-list 101 permit ip host 10.0.2.30 host 10.0.2.31 access-list 101 deny udp any eq rip any access-list 101 deny udp any any eq rip access-list 101 deny udp any eq isakmp any access-list 101 deny udp any any eq isakmp access-list 101 permit ip any any snmp-server engineID local 000000090200003094202740 snmp-server community public RW ! line con 0 exec-timeout 0 0 transport input none line aux 0 line vty 0 4 password xxx login ! end
The following example illustrates how to configure call transfer. In Figure 1, User A and User C are in an established call. User C then transfers the call to User B. This results in call establishment between User A and User B. User C is then disconnected with User A, regardless of whether the transfer fails or succeeds.
When a call originates or terminates on a gateway, either the calling party number, the called party number, or the port is used (depending on the scenario) to match a dial peer in order to determine the basic call characteristics. One of the characteristics to determine is which application to use for the call. For the call transfer to succeed, the matching dial peer must have application set to "session" on the gateway that is controlling the transfer. (This is the gateway that receives the Bye with an Also header).
There are two scenarios for dial-peer matching based on whether the call is coming from a POTS interface or from the IP network.
a. The called number matches the "incoming called-number" on a VoIP dial peer.
b. The calling number matches the "answer-address" on a VoIP dial peer.
c. The calling number matches the "destination-pattern" on a VoIP dial peer.
 |
Note For calls coming from the IP network, it is possible for the calling number to be blocked based on privacy restrictions. In such cases, the "incoming called-number" be used for call transfers. |
In this example, Gateway 1 handles the transfer (recipient of the Bye with the Also header). User C invokes the transfer service (originator of the Bye with the Also header). There are two scenarios in which a dial peer match must have application set to "session" for the transfer to succeed:
|
Note To handle all call transfer situations, you should configure both POTS and VoIP dial peers. |
To configure a POTS dial peer with the application of session, do the following:
| Step | Command | Purpose |
|---|---|---|
1. | Router (config-dial-peer) | Enter the dial-peer mode to configure a POTS dial peer. |
2. | Router (config-dial-peer) # application session | Specifies that the standard session application will be invoked for this dial peer. |
To configure a VoIP dial peer with a destination pattern, do the following:
| Step | Command | Purpose |
|---|---|---|
1. | Router (config-dial-peer) | Enter the dial-peer mode to configure a VoIP dial peer. |
2. | Router (config-dial-peer) # destination-pattern pattern | Specifies the telephone number associated with the dial peer. |
To configure a VoIP dial peer with an incoming called-number, do the following:
| Step | Command | Purpose |
|---|---|---|
1. | Router (config-dial-peer) | Enter the dial-peer mode to configure a VoIP dial peer. |
2. | Router (config-dial-peer) # incoming called-number number | Specifies an incoming called number of a dial peer. |
To configure a VoIP dial peer with an incoming called-number, do the following:
| Step | Command | Purpose |
|---|---|---|
1. | Router (config-dial-peer) | Enter the dial-peer mode to configure a VoIP dial peer. |
2. | Router (config-dial-peer) # answer-address [+]string[T] | Specifies the full E.164 telephone number to be used to identify the dial peer of an incoming call. |
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 the following commands:
Enter the default command to reset the value of a SIP-related command to its default.
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, which means that tones are fed from the terminating gateway. 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
12.1(3)T This command was introduced.
Release
Modification
Examples
Router (config-sip-ua)# default inband-alerting
Enter the gw-accounting command, to enable gateway-specific accounting. There are three different methods of accounting:
To disable gateway-specific accounting, use the no form of this command.
gw-accounting { voip | syslog | h323 [syslog] }
Syntax Description
voip Uses RADIUS to output accounting CDRs. Both H.323 and SIP protocols can use this method, so the name is not bound to a protocol. syslog Uses the system logging facility to output CDRs. h323 Uses RADIUS to output accounting CDRs.
Defaults
There are no default behaviors or values for this command.
Command Modes
Global configuration
Command History
11.3(6)NA2 This command was introduced. 12.1(1)T The voip option was added.
Release
Modification
Usage Guidelines
Examples
Router (config)# gw-accounting voip
Enter the inband-alerting command in the SIP user agent configuration mode to enable inband alerting. Use the no form of this command to disable inband alerting. If inband alerting is enabled, the originating gateway can open an early media path (upon receiving a 180 or 183 message with a SDP body). This allows the terminating gateway or switch to feed tones or announcements before the call is connected. If inband-alerting is disabled, local alerting is generated on the originating gateway.
[no] inband-alertingSyntax Description
There are no parameters for this command.
Defaults
By default, inband alerting is enabled.
Command Modes
SIP user agent configuration
Command History
12.1(1)T This command was introduced. 12.1(3)T This command was limited to enabling and disabling inband alerting.
Release
Modification
Usage Guidelines
To reset this command to the default value, use the default command.
Examples
Router (config)# sip-ua Router (config-sip-ua)# no inband-alerting
Enter the max-forwards command in the SIP user agent configuration mode to set the maximum number of proxy or redirect servers that can forward the request. To reset this command to the default value, use the no form of the command.
max-forwards number
Syntax Description
number Number of hops. Possible values are 1 through 15. The default is 6.
Defaults
The default number of hops is 6.
Command Modes
SIP user agent configuration
Command History
12.1(3)T This command was introduced.
Release
Modification
Usage Guidelines
To reset this command to the default value, you can also use the default command.
Examples
Router (config)# sip-ua Router (config-sip-ua)# max-forwards 2
Enter the max-redirects command in the dial-peer configuration mode to set the maximum number of redirect servers that a call can traverse. To reset this command to the default value, use the no form of the command.
max-redirects number
Syntax Description
number Maximum number of redirect servers that a call can traverse. Possible values are 1 through 10. The default is 1.
Defaults
The default number of redirects is 1.
Command Modes
dial-peer configuration
Command History
12.1(1)T This command was introduced.
Release
Modification
Examples
Router (config)# dial-peer voice 102 voip Router (config-dial-peer)# max-redirects 2
Enter the retry command in the SIP user agent configuration mode to configure the maximum number of retry attempts for SIP messages. 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. Possible values are 1 through 10. The default is 6. response number Number of RESPONSE retries. Possible values are 1 through 10. The default is 6. bye number Number of BYE retries. Possible values are 1 through 10. The default is 10. cancel number Number of CANCEL retries. Possible values are 1 through 10. The default is 10.
Defaults
The default number of retries for invite and response is 6, for bye and cancel is 10.
Command Modes
SIP user agent configuration
Command History
12.1(1)T This command was introduced.
Release
Modification
Usage Guidelines
To reset this command to the default value, you can also use the default command.
Examples
Router (config)# sip-ua Router (config-sip-ua)# retry invite 5
Enter the session protocol command in the dial-peer configuration mode to configure a VoIP dial peer to use either H323 or SIP as the session protocol for VoIP call signaling. 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 Configures the dial peer to use IETF SIP. SIP users should use this new option.
Defaults
There are no default behaviors or values for this command.
Command Modes
dial-peer configuration
Command History
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.
Release
Modification
Examples
Router (config)# dial-peer voice 102 voip Router (config-dial-peer)# session protocol sipv2
Enter the session target command to specify a network-specific address for a dial peer. To reset this command to the default value, use the no form of this command.
session target { sip-server | dns:[$s$. | $d$. | $e$. | $u$.]host-name | ipv4:ip_addr[:port-num] | ras}
Syntax Description
sip-server 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 format: name.gateway.xyz (Optional) You can use one of the following wildcards with this keyword when defining the session target for VoIP peers: ipv4:ip_addr Sets the IP address of the dial peer. A valid IP address is in this format: xxx.xxx.xxx.xxx port-num (Optional) The UDP port number to use 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. This option is not valid for SIP.
Defaults
The default for this command is that the session target is enabled but no IP address or domain name is defined.
Command Modes
Dial-peer configuration
Command History
11.3(1)T This command was introduced. 12.1(1)T The sip-server option was added.
Release
Modification
Usage Guidelines
Examples
Router (config)# dial-peer voice 102 voip Router (config-dial-peer)# session target dns:UA-1-f0.sip.com
Enter the session transport command to configure the VoIP dial peer to use TCP or UDP as the underlying transport layer protocol for SIP messages. To reset the value of this command to the default, 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. This is the default. tcp Configure the SIP dial peer to use the TCP transport layer protocol.
Defaults
The default for this command is that the SIP dial peer uses UDP.
|
Note The transport protocol specified with the transport command and the one specified with the session transport command must be the same. |
Command Modes
Dial-peer configuration
Command History
12.1(1)T This command was introduced.
Release
Modification
Usage Guidelines
Use show sip-ua status to ensure that the transport protocol that you set using the session transport command matches the protocol set using the transport command.
Examples
Router (config)# dial-peer voice 102 voip Router (dial-peer-config)# session transport udp
Enter the show sip-ua statistics command to display response, traffic, and retry SIP statistics.
show sip-ua statisticsSyntax Description
This command has no arguments or keywords.
Defaults
There are no default behaviors or values for this command.
Command Modes
EXEC
Command History
12.1(3)T This command was introduced.
Release
Modification
Examples
Router#show sip-ua statistics
SIP Response Statistics (Inbound/Outbound)
Informational:
Trying 0/0, Ringing 0/0,
Forwarded 0/0, Queued 0/0,
SessionProgress 0/0
Success:
OkInvite 0/0, OkBye 0/0,
OkCancel 0/0, OkOptions 0/0
Redirection (Inbound only):
MultipleChoice 0, MovedPermanently 0,
MovedTemporarily 0, SeeOther 0,
UseProxy 0, AlternateService 0
Client Error:
BadRequest 0/0, Unauthorized 0/0,
PaymentRequired 0/0, Forbidden 0/0,
NotFound 0/0, MethodNotAllowed 0/0,
NotAcceptable 0/0, ProxyAuthReqd 0/0,
ReqTimeout 0/0, Conflict 0/0, Gone 0/0,
LengthRequired 0/0, ReqEntityTooLarge 0/0,
ReqURITooLarge 0/0, UnsupportedMediaType 0/0,
BadExtension 0/0, TempNotAvailable 0/0,
CallLegNonExistent 0/0, LoopDetected 0/0,
TooManyHops 0/0, AddrIncomplete 0/0,
Ambiguous 0/0, BusyHere 0/0
Server Error:
InternalError 0/0, NotImplemented 0/0,
BadGateway 0/0, ServiceUnavail 0/0,
GatewayTimeout 0/0, BadSipVer 0/0
Global Failure:
BusyEverywhere 0/0, Decline 0/0,
NoExistAnywhere 0/0, NotAcceptable 0/0
SIP Total Traffic Statistics (Inbound/Outbound)
Invite 0/0, Ack 0/0, Bye 0/0,
Cancel 0/0, Options 0/0
Retry Statistics
Invite 0, Bye 0, Cancel 0, Response 0
Enter the show sip-ua status command to display SIP status.
show sip-ua statusSyntax Description
This command has no arguments or keywords.
Defaults
There are no default behaviors or values for this command.
Command Modes
EXEC
Command History
12.1(1)T This command was introduced. 12.1(3)T The statistics portion of the output was removed and is now included in the show sip-ua statistics command.
Release
Modification
Examples
Router#show sip-ua status SIP User Agent Status SIP User Agent for UDP : ENABLED SIP User Agent for TCP : ENABLED SIP max-forwards :6
Enter the show sip-ua timers command to display the current settings for SIP UA timers.
show sip-ua timersSyntax Description
This command has no arguments or keywords.
Defaults
There are no default behaviors or values for this command.
Command Modes
EXEC
Command History
12.1(1)T This command was introduced. 12.1(3)T The output of this command was changed to reflect the changes in the timers command.
Release
Modification
Examples
Router#show sip-ua timers SIP UA Timer Values (millisecs) trying 500, expires 180000, connect 500, disconnect 500
Enter the sip-server command in the SIP user agent configuration mode to configure a network address for the SIP server interface.
sip-server { dns:[host-name] | ipv4:ipaddr[:port-num] }
Syntax Description
dns: Sets the global SIP server interface to a DNS host name. If you do not specify a host name, the default DNS defined by the ip name-server command is used. host-name A valid DNS host name takes the following format: name.gateway.xyz ipv4:ip_addr Sets the global SIP server interface to an IP address. A valid IP address takes the following format: xxx.xxx.xxx.xxx port-num (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
12.1(1)T This command was introduced.
Release
Modification
Usage Guidelines
Examples
Router (config)# sip-ua Router (config-sip-ua)# sip-server dns:UA-1-f0.sip.com
Enter the sip-ua command in the global configuration mode to enable the sip-ua configuration commands, with which you configure the user agent. 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
12.1(1)T This command was introduced.
Release
Modification
Usage Guidelines
Enter the sip-ua command to enter the SIP user agent-configuration sub-mode. The sub-mode configuration commands are:
exit | Exits the SIP user agent configuration mode. |
Specifies an inband-alerting SIP header. | |
max-forwards | Specifies the maximum number of hops for a request. |
Configures the SIP signaling timers for retry attempts. | |
Configures a SIP server interface. | |
Configures the SIP signaling timers configuration. | |
Enables/disables a SIP UA transport for TCP/UDP. |
Examples
Router (config)#sip-ua Router (config-sip-ua)#? SIP UA configuration commands: default Set a command to its defaults exit Exit from sip-ua configuration mode inband-alerting Specify an Inband-alerting SIP header max-forwards Change number of max-forwards for SIP Methods no Negate a command or set its defaults retry Change default retries for each SIP Method sip-server Configure a SIP Server Interface timers SIP Signaling Timers Configuration transport Enable SIP UA transport for TCP/UDP
Enter the timers command in the SIP user agent configuration mode to configure the SIP signaling timers. To reset this command to the default value, use the no form of this command.
timers { trying number | connect number | disconnect number | expires number }
Syntax Description
trying number Time (in milliseconds) to wait for a 100 response to an INVITE request. Possible values are 100 through 1000. The default is 500. connect number Time (in milliseconds) to wait for a 200 response to an ACK request. Possible values are 100 through 1000. The default is 500. disconnect number Time (in milliseconds) to wait for a 200 response to a BYE request. Possible values are 100 through 1000. The default is 500. expires number Time (in milliseconds) for which an INVITE request is valid. Possible values are 60000 through 300000. The default is 180000.
Defaults
The default for trying, connect, and disconnect is 500. The default for expires is 180000.
Command Modes
SIP user agent configuration
Command History
12.1(1)T This command was introduced. 12.1(3)T This command was modified. The names of the parameters were changed. Two of the parameters (invite-wait-180 and invite-wait-200) were combined into one (trying).
Release
Modification
Usage Guidelines
Examples
Router (config)# sip-ua Router (config-sip-ua)# timers trying 500
Enter the transport command in the SIP user agent configuration mode to configure the SIP user agent (gateway) for SIP signaling messages on inbound calls through the SIP TCP or UDP socket.
This command controls whether messages reach the SIP service provider interface (SPI). To block reception of SIP signaling messages on a particular 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, both UDP and TCP transport protocols are enabled.
Command Modes
SIP user agent configuration
Command History
12.1(1)T This command was introduced.
Release
Modification
Usage Guidelines
To reset this command to the default value, use the default command.
Examples
Router (config)# sip-ua Router (config-sip-ua)# no transport tcp
This section documents new and modified debug commands associated with the SIP feature.
Enter the debug ccsip all command to enable all SIP-related debugging. To disable all debugging output, use the no form of this command.
debug ccsip allSyntax Description
This command has no arguments or keywords.
Command Modes
EXEC
Command History
12.1(1)T This command was introduced. 12.1.(3)T The output of the command was changed.
Release
Modification
Usage Guidelines
The debug ccsip all command enables the following debug SIP commands:
Examples
From one side of the call, the debug output is as follows:
Router1#debug ccsip all
All SIP call tracing enabled
Router1#
*Mar 6 14:10:42: 0x624CFEF8 : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)
*Mar 6 14:10:42: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP
*Mar 6 14:10:42: CCSIP-SPI-CONTROL: act_idle_call_setup
*Mar 6 14:10:42: act_idle_call_setup:Not using Voice Class Codec
*Mar 6 14:10:42: act_idle_call_setup: preferred_codec set[0] type :g711ulaw bytes: 160
*Mar 6 14:10:42: Queued event from SIP SPI : SIPSPI_EV_CREATE_CONNECTION
*Mar 6 14:10:42: 0x624CFEF8 : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_CONNECTING)
*Mar 6 14:10:42: REQUEST CONNECTION TO IP:166.34.245.231 PORT:5060
*Mar 6 14:10:42: 0x624CFEF8 : State change from (STATE_IDLE, SUBSTATE_CONNECTING) to (STATE_IDLE, SUBSTATE_CONNECTING)
*Mar 6 14:10:42: CCSIP-SPI-CONTROL: act_idle_connection_created
*Mar 6 14:10:42: CCSIP-SPI-CONTROL: act_idle_connection_created: Connid(1) created to 166.34.245.231:5060, local_port 54113
*Mar 6 14:10:42: sipSPIAddLocalContact
*Mar 6 14:10:42: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
*Mar 6 14:10:42: CCSIP-SPI-CONTROL: sip_stats_method
*Mar 6 14:10:42: 0x624CFEF8 : State change from (STATE_IDLE, SUBSTATE_CONNECTING) to (STATE_SENT_INVITE, SUBSTATE_NONE)
*Mar 6 14:10:42: Sent:
INVITE sip:3660210@166.34.245.231;user=phone;phone-context=unknown SIP/2.0
Via: SIP/2.0/UDP 166.34.245.230:54113
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>
Date: Sat, 06 Mar 1993 19:10:42 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
Cisco-Guid: 2881152943-2184249548-0-483039712
User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq: 101 INVITE
Max-Forwards: 6
Timestamp: 731427042
Contact: <sip:3660110@166.34.245.230:5060;user=phone>
Expires: 180
Content-Type: application/sdp
Content-Length: 137
v=0
o=CiscoSystemsSIP-GW-UserAgent 1212 283 IN IP4 166.34.245.230
s=SIP Call
t=0 0
c=IN IP4 166.34.245.230
m=audio 20208 RTP/AVP 0
*Mar 6 14:10:42: Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 166.34.245.230:54113
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>
Date: Mon, 08 Mar 1993 22:36:40 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
Timestamp: 731427042
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq: 101 INVITE
Content-Length: 0
*Mar 6 14:10:42: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.231:5060
*Mar 6 14:10:42: CCSIP-SPI-CONTROL: act_sentinvite_new_message
*Mar 6 14:10:42: CCSIP-SPI-CONTROL: sipSPICheckResponse
*Mar 6 14:10:42: CCSIP-SPI-CONTROL: sip_stats_status_code
*Mar 6 14:10:42: Roundtrip delay 4 milliseconds for method INVITE
*Mar 6 14:10:42: 0x624CFEF8 : State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)
*Mar 6 14:10:42: Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 166.34.245.230:54113
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>
Date: Mon, 08 Mar 1993 22:36:40 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
Timestamp: 731427042
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq: 101 INVITE
Content-Type: application/sdp
Content-Length: 137
v=0
o=CiscoSystemsSIP-GW-UserAgent 969 7889 IN IP4 166.34.245.231
s=SIP Call
t=0 0
c=IN IP4 166.34.245.231
m=audio 20038 RTP/AVP 0
*Mar 6 14:10:42: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.231:5060
*Mar 6 14:10:42: CCSIP-SPI-CONTROL: act_recdproc_new_message
*Mar 6 14:10:42: CCSIP-SPI-CONTROL: sipSPICheckResponse
*Mar 6 14:10:42: CCSIP-SPI-CONTROL: sipSPICheckResponse : Updating session description
*Mar 6 14:10:42: CCSIP-SPI-CONTROL: sip_stats_status_code
*Mar 6 14:10:42: Roundtrip delay 8 milliseconds for method INVITE
*Mar 6 14:10:42: HandleSIP1xxRinging: SDP MediaTypes negotiation successful!
Negotiated Codec : g711ulaw , bytes :160
Inband Alerting : 0
*Mar 6 14:10:42: 0x624CFEF8 : State change from (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_ALERTING)
*Mar 6 14:10:46: Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 166.34.245.230:54113
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27D3FCA8-C7F
Date: Mon, 08 Mar 1993 22:36:40 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
Timestamp: 731427042
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Contact: <sip:3660210@166.34.245.231:5060;user=phone>
CSeq: 101 INVITE
Content-Type: application/sdp
Content-Length: 137
v=0
o=CiscoSystemsSIP-GW-UserAgent 969 7889 IN IP4 166.34.245.231
s=SIP Call
t=0 0
c=IN IP4 166.34.245.231
m=audio 20038 RTP/AVP 0
*Mar 6 14:10:46: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.231:5060
*Mar 6 14:10:46: CCSIP-SPI-CONTROL: act_recdproc_new_message
*Mar 6 14:10:46: CCSIP-SPI-CONTROL: sipSPICheckResponse
*Mar 6 14:10:46: CCSIP-SPI-CONTROL: sipSPICheckResponse : Updating session description
*Mar 6 14:10:46: CCSIP-SPI-CONTROL: sip_stats_status_code
*Mar 6 14:10:46: Roundtrip delay 3536 milliseconds for method INVITE
*Mar 6 14:10:46: CCSIP-SPI-CONTROL: act_recdproc_new_message: SDP MediaTypes negotiation successful!
Negotiated Codec : g711ulaw , bytes :160
*Mar 6 14:10:46: CCSIP-SPI-CONTROL: sipSPIReconnectConnection
*Mar 6 14:10:46: Queued event from SIP SPI : SIPSPI_EV_RECONNECT_CONNECTION
*Mar 6 14:10:46: CCSIP-SPI-CONTROL: recv_200_OK_for_invite
*Mar 6 14:10:46: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
*Mar 6 14:10:46: CCSIP-SPI-CONTROL: sip_stats_method
*Mar 6 14:10:46: 0x624CFEF8 : State change from (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_ALERTING) to (STATE_ACTIVE, SUBSTATE_NONE)
*Mar 6 14:10:46: The Call Setup Information is :
Call Control Block (CCB) : 0x624CFEF8
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 3660110
Called Number : 3660210
Negotiated Codec : g711ulaw
Source IP Address (Media): 166.34.245.230
Source IP Port (Media): 20208
Destn IP Address (Media): 166.34.245.231
Destn IP Port (Media): 20038
Destn SIP Addr (Control) : 166.34.245.231
Destn SIP Port (Control) : 5060
Destination Name : 166.34.245.231
*Mar 6 14:10:46: HandleUdpReconnection: Udp socket connected for fd: 1 with 166.34.245.231:5060
*Mar 6 14:10:46: Sent:
ACK sip:3660210@166.34.245.231:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 166.34.245.230:54113
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27D3FCA8-C7F
Date: Sat, 06 Mar 1993 19:10:42 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
Max-Forwards: 6
Content-Type: application/sdp
Content-Length: 137
CSeq: 101 ACK
v=0
o=CiscoSystemsSIP-GW-UserAgent 1212 283 IN IP4 166.34.245.230
s=SIP Call
t=0 0
c=IN IP4 166.34.245.230
m=audio 20208 RTP/AVP 0
*Mar 6 14:10:46: CCSIP-SPI-CONTROL: ccsip_caps_ind
*Mar 6 14:10:46: ccsip_caps_ind: Load DSP with codec (5) g711ulaw, Bytes=160
*Mar 6 14:10:46: ccsip_caps_ind: set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_INBAND_VOICE
*Mar 6 14:10:46: CCSIP-SPI-CONTROL: ccsip_caps_ack
*Mar 6 14:10:50: Received:
BYE sip:3660110@166.34.245.230:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 166.34.245.231:54835
From: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27D3FCA8-C7F
To: "3660110" <sip:3660110@166.34.245.230>
Date: Mon, 08 Mar 1993 22:36:44 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Max-Forwards: 6
Timestamp: 731612207
CSeq: 101 BYE
Content-Length: 0
*Mar 6 14:10:50: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.231:54835
*Mar 6 14:10:50: CCSIP-SPI-CONTROL: act_active_new_message
*Mar 6 14:10:50: CCSIP-SPI-CONTROL: sact_active_new_message_request
*Mar 6 14:10:50: CCSIP-SPI-CONTROL: sip_stats_method
*Mar 6 14:10:50: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
*Mar 6 14:10:50: CCSIP-SPI-CONTROL: sip_stats_status_code
*Mar 6 14:10:50: CCSIP-SPI-CONTROL: sipSPIInitiateCallDisconnect : Initiate call disconnect(16) for outgoing call
*Mar 6 14:10:50: 0x624CFEF8 : State change from (STATE_ACTIVE, SUBSTATE_NONE) to (STATE_DISCONNECTING, SUBSTATE_NONE)
*Mar 6 14:10:50: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 166.34.245.231:54835
From: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27D3FCA8-C7F
To: "3660110" <sip:3660110@166.34.245.230>
Date: Sat, 06 Mar 1993 19:10:50 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Timestamp: 731612207
Content-Length: 0
CSeq: 101 BYE
*Mar 6 14:10:50: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_DISCONNECT
*Mar 6 14:10:50: CCSIP-SPI-CONTROL: act_disconnecting_disconnect
*Mar 6 14:10:50: CCSIP-SPI-CONTROL: sipSPICallCleanup
*Mar 6 14:10:50: Queued event from SIP SPI : SIPSPI_EV_CLOSE_CONNECTION
*Mar 6 14:10:50: CLOSE CONNECTION TO CONNID:1
*Mar 6 14:10:50: sipSPIIcpifUpdate :CallState: 4 Playout: 1755 DiscTime:48305031 ConnTime 48304651
*Mar 6 14:10:50: 0x624CFEF8 : State change from (STATE_DISCONNECTING, SUBSTATE_NONE) to (STATE_DEAD, SUBSTATE_NONE)
*Mar 6 14:10:50: The Call Setup Information is :
Call Control Block (CCB) : 0x624CFEF8
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 3660110
Called Number : 3660210
Negotiated Codec : g711ulaw
Source IP Address (Media): 166.34.245.230
Source IP Port (Media): 20208
Destn IP Address (Media): 166.34.245.231
Destn IP Port (Media): 20038
Destn SIP Addr (Control) : 166.34.245.231
Destn SIP Port (Control) : 5060
Destination Name : 166.34.245.231
*Mar 6 14:10:50:
Disconnect Cause (CC) : 16
Disconnect Cause (SIP) : 200
*Mar 6 14:10:50: udpsock_close_connect: Socket fd: 1 closed for connid 1 with remote port: 5060
Router1#
From the other side of the call, the debug output is as follows:
3660-2#debug ccsip all
All SIP call tracing enabled
3660-2#
*Mar 8 17:36:40: Received:
INVITE sip:3660210@166.34.245.231;user=phone;phone-context=unknown SIP/2.0
Via: SIP/2.0/UDP 166.34.245.230:54113
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>
Date: Sat, 06 Mar 1993 19:10:42 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
Cisco-Guid: 2881152943-2184249548-0-483039712
User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq: 101 INVITE
Max-Forwards: 6
Timestamp: 731427042
Contact: <sip:3660110@166.34.245.230:5060;user=phone>
Expires: 180
Content-Type: application/sdp
Content-Length: 137
v=0
o=CiscoSystemsSIP-GW-UserAgent 1212 283 IN IP4 166.34.245.230
s=SIP Call
t=0 0
c=IN IP4 166.34.245.230
m=audio 20208 RTP/AVP 0
*Mar 8 17:36:40: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.230:54113
*Mar 8 17:36:40: CCSIP-SPI-CONTROL: sipSPISipIncomingCall
*Mar 8 17:36:40: 0x624D8CCC : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)
*Mar 8 17:36:40: CCSIP-SPI-CONTROL: act_idle_new_message
*Mar 8 17:36:40: CCSIP-SPI-CONTROL: sact_idle_new_message_invite
*Mar 8 17:36:40: CCSIP-SPI-CONTROL: sip_stats_method
*Mar 8 17:36:40: sact_idle_new_message_invite:Not Using Voice Class Codec
*Mar 8 17:36:40: sact_idle_new_message_invite: Preferred codec[0] type: g711ulaw Bytes :160
*Mar 8 17:36:40: sact_idle_new_message_invite: Media Negotiation successful for an
incoming call
*Mar 8 17:36:40: sact_idle_new_message_invite: Negotiated Codec : g711ulaw, bytes :160
Preferred Codec : g711ulaw, bytes :160
*Mar 8 17:36:40: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
*Mar 8 17:36:40: CCSIP-SPI-CONTROL: sip_stats_status_code
*Mar 8 17:36:40: Num of Contact Locations 1 3660110 166.34.245.230 5060
*Mar 8 17:36:40: 0x624D8CCC : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_RECD_INVITE, SUBSTATE_RECD_INVITE_CALL_SETUP)
*Mar 8 17:36:40: Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 166.34.245.230:54113
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>
Date: Mon, 08 Mar 1993 22:36:40 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
Timestamp: 731427042
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq: 101 INVITE
Content-Length: 0
*Mar 8 17:36:40: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_PROCEEDING
*Mar 8 17:36:40: CCSIP-SPI-CONTROL: act_recdinvite_proceeding
*Mar 8 17:36:40: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_ALERTING
*Mar 8 17:36:40: CCSIP-SPI-CONTROL: ccsip_caps_ind
*Mar 8 17:36:40: ccsip_caps_ind: codec(negotiated) = 5(Bytes 160)
*Mar 8 17:36:40: ccsip_caps_ind: Load DSP with codec (5) g711ulaw, Bytes=160
*Mar 8 17:36:40: ccsip_caps_ind: set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_INBAND_VOICE
*Mar 8 17:36:40: CCSIP-SPI-CONTROL: ccsip_caps_ack
*Mar 8 17:36:40: CCSIP-SPI-CONTROL: act_recdinvite_alerting
*Mar 8 17:36:40: 180 Ringing with SDP - not likely
*Mar 8 17:36:40: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
*Mar 8 17:36:40: CCSIP-SPI-CONTROL: sip_stats_status_code
*Mar 8 17:36:40: 0x624D8CCC : State change from (STATE_RECD_INVITE, SUBSTATE_RECD_INVITE_CALL_SETUP) to (STATE_SENT_ALERTING, SUBSTATE_NONE)
*Mar 8 17:36:40: Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 166.34.245.230:54113
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>
Date: Mon, 08 Mar 1993 22:36:40 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
Timestamp: 731427042
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq: 101 INVITE
Content-Type: application/sdp
Content-Length: 137
v=0
o=CiscoSystemsSIP-GW-UserAgent 969 7889 IN IP4 166.34.245.231
s=SIP Call
t=0 0
c=IN IP4 166.34.245.231
m=audio 20038 RTP/AVP 0
*Mar 8 17:36:44: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_CONNECT
*Mar 8 17:36:44: CCSIP-SPI-CONTROL: act_sentalert_connect
*Mar 8 17:36:44: sipSPIAddLocalContact
*Mar 8 17:36:44: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
*Mar 8 17:36:44: CCSIP-SPI-CONTROL: sip_stats_status_code
*Mar 8 17:36:44: 0x624D8CCC : State change from (STATE_SENT_ALERTING, SUBSTATE_NONE) to (STATE_SENT_SUCCESS, SUBSTATE_NONE)
*Mar 8 17:36:44: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 166.34.245.230:54113
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27D3FCA8-C7F
Date: Mon, 08 Mar 1993 22:36:40 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
Timestamp: 731427042
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Contact: <sip:3660210@166.34.245.231:5060;user=phone>
CSeq: 101 INVITE
Content-Type: application/sdp
Content-Length: 137
v=0
o=CiscoSystemsSIP-GW-UserAgent 969 7889 IN IP4 166.34.245.231
s=SIP Call
t=0 0
c=IN IP4 166.34.245.231
m=audio 20038 RTP/AVP 0
*Mar 8 17:36:44: Received:
ACK sip:3660210@166.34.245.231:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 166.34.245.230:54113
From: "3660110" <sip:3660110@166.34.245.230>
To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27D3FCA8-C7F
Date: Sat, 06 Mar 1993 19:10:42 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
Max-Forwards: 6
Content-Type: application/sdp
Content-Length: 137
CSeq: 101 ACK
v=0
o=CiscoSystemsSIP-GW-UserAgent 1212 283 IN IP4 166.34.245.230
s=SIP Call
t=0 0
c=IN IP4 166.34.245.230
m=audio 20208 RTP/AVP 0
*Mar 8 17:36:44: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.230:54113
*Mar 8 17:36:44: CCSIP-SPI-CONTROL: act_sentsucc_new_message
*Mar 8 17:36:44: CCSIP-SPI-CONTROL: sip_stats_method
*Mar 8 17:36:44: 0x624D8CCC : State change from (STATE_SENT_SUCCESS, SUBSTATE_NONE) to (STATE_ACTIVE, SUBSTATE_NONE)
*Mar 8 17:36:44: The Call Setup Information is :
Call Control Block (CCB) : 0x624D8CCC
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 3660110
Called Number : 3660210
Negotiated Codec : g711ulaw
Source IP Address (Media): 166.34.245.231
Source IP Port (Media): 20038
Destn IP Address (Media): 166.34.245.230
Destn IP Port (Media): 20208
Destn SIP Addr (Control) : 166.34.245.230
Destn SIP Port (Control) : 5060
Destination Name : 166.34.245.230
*Mar 8 17:36:47: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_DISCONNECT
*Mar 8 17:36:47: CCSIP-SPI-CONTROL: act_active_disconnect
*Mar 8 17:36:47: Queued event from SIP SPI : SIPSPI_EV_CREATE_CONNECTION
*Mar 8 17:36:47: 0x624D8CCC : State change from (STATE_ACTIVE, SUBSTATE_NONE) to (STATE_ACTIVE, SUBSTATE_CONNECTING)
*Mar 8 17:36:47: REQUEST CONNECTION TO IP:166.34.245.230 PORT:5060
*Mar 8 17:36:47: 0x624D8CCC : State change from (STATE_ACTIVE, SUBSTATE_CONNECTING) to (STATE_ACTIVE, SUBSTATE_CONNECTING)
*Mar 8 17:36:47: CCSIP-SPI-CONTROL: act_active_connection_created
*Mar 8 17:36:47: CCSIP-SPI-CONTROL: sipSPICheckSocketConnection
*Mar 8 17:36:47: CCSIP-SPI-CONTROL: sipSPICheckSocketConnection: Connid(1) created to 166.34.245.230:5060, local_port 54835
*Mar 8 17:36:47: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
*Mar 8 17:36:47: CCSIP-SPI-CONTROL: sip_stats_method
*Mar 8 17:36:47: 0x624D8CCC : State change from (STATE_ACTIVE, SUBSTATE_CONNECTING) to (STATE_DISCONNECTING, SUBSTATE_NONE)
*Mar 8 17:36:47: Sent:
BYE sip:3660110@166.34.245.230:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 166.34.245.231:54835
From: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27D3FCA8-C7F
To: "3660110" <sip:3660110@166.34.245.230>
Date: Mon, 08 Mar 1993 22:36:44 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Max-Forwards: 6
Timestamp: 731612207
CSeq: 101 BYE
Content-Length: 0
*Mar 8 17:36:47: Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 166.34.245.231:54835
From: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27D3FCA8-C7F
To: "3660110" <sip:3660110@166.34.245.230>
Date: Sat, 06 Mar 1993 19:10:50 GMT
Call-ID: ABBAE7AF-823100CE-0-1CCAA69C@172.18.192.194
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Timestamp: 731612207
Content-Length: 0
CSeq: 101 BYE
*Mar 8 17:36:47: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.230:54113
*Mar 8 17:36:47: CCSIP-SPI-CONTROL: act_disconnecting_new_message
*Mar 8 17:36:47: CCSIP-SPI-CONTROL: sact_disconnecting_new_message_response
*Mar 8 17:36:47: CCSIP-SPI-CONTROL: sipSPICheckResponse
*Mar 8 17:36:47: CCSIP-SPI-CONTROL: sip_stats_status_code
*Mar 8 17:36:47: Roundtrip delay 4 milliseconds for method BYE
*Mar 8 17:36:47: CCSIP-SPI-CONTROL: sipSPICallCleanup
*Mar 8 17:36:47: Queued event from SIP SPI : SIPSPI_EV_CLOSE_CONNECTION
*Mar 8 17:36:47: CLOSE CONNECTION TO CONNID:1
*Mar 8 17:36:47: sipSPIIcpifUpdate :CallState: 4 Playout: 1265 DiscTime:66820800 ConnTime 66820420
*Mar 8 17:36:47: 0x624D8CCC : State change from (STATE_DISCONNECTING, SUBSTATE_NONE) to (STATE_DEAD, SUBSTATE_NONE)
*Mar 8 17:36:47: The Call Setup Information is :
Call Control Block (CCB) : 0x624D8CCC
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 3660110
Called Number : 3660210
Negotiated Codec : g711ulaw
Source IP Address (Media): 166.34.245.231
Source IP Port (Media): 20038
Destn IP Address (Media): 166.34.245.230
Destn IP Port (Media): 20208
Destn SIP Addr (Control) : 166.34.245.230
Destn SIP Port (Control) : 5060
Destination Name : 166.34.245.230
*Mar 8 17:36:47:
Disconnect Cause (CC) : 16
Disconnect Cause (SIP) : 200
*Mar 8 17:36:47: udpsock_close_connect: Socket fd: 1 closed for connid 1 with remote port: 5060
Enter the debug ccsip calls command to show all SIP Service Provider Interface (SPI) call tracing. This command traces the SIP call details as they are updated in the SIP call control block.
debug ccsip callsSyntax Description
This command has no arguments or keywords.
Command Modes
EXEC
Command History
12.1(1)T This command was introduced. 12.1.(3)T The output of the command was changed.
Release
Modification
Examples
From one side of the call, the debug output is as follows:
Router1#debug ccsip calls
SIP Call statistics tracing is enabled
Router1#
*Mar 6 14:12:33: The Call Setup Information is :
Call Control Block (CCB) : 0x624D078C
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 3660110
Called Number : 3660210
Negotiated Codec : g711ulaw
Source IP Address (Media): 166.34.245.230
Source IP Port (Media): 20644
Destn IP Address (Media): 166.34.245.231
Destn IP Port (Media): 20500
Destn SIP Addr (Control) : 166.34.245.231
Destn SIP Port (Control) : 5060
Destination Name : 166.34.245.231
*Mar 6 14:12:40: The Call Setup Information is :
Call Control Block (CCB) : 0x624D078C
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 3660110
Called Number : 3660210
Negotiated Codec : g711ulaw
Source IP Address (Media): 166.34.245.230
Source IP Port (Media): 20644
Destn IP Address (Media): 166.34.245.231
Destn IP Port (Media): 20500
Destn SIP Addr (Control) : 166.34.245.231
Destn SIP Port (Control) : 5060
Destination Name : 166.34.245.231
*Mar 6 14:12:40:
Disconnect Cause (CC) : 16
Disconnect Cause (SIP) : 200
Router1#
From the other side of the call, the debug output is as follows:
Router2#debug ccsip calls
SIP Call statistics tracing is enabled
Router2#
*Mar 8 17:38:31: The Call Setup Information is :
Call Control Block (CCB) : 0x624D9560
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 3660110
Called Number : 3660210
Negotiated Codec : g711ulaw
Source IP Address (Media): 166.34.245.231
Source IP Port (Media): 20500
Destn IP Address (Media): 166.34.245.230
Destn IP Port (Media): 20644
Destn SIP Addr (Control) : 166.34.245.230
Destn SIP Port (Control) : 5060
Destination Name : 166.34.245.230
*Mar 8 17:38:38: The Call Setup Information is :
Call Control Block (CCB) : 0x624D9560
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 3660110
Called Number : 3660210
Negotiated Codec : g711ulaw
Source IP Address (Media): 166.34.245.231
Source IP Port (Media): 20500
Destn IP Address (Media): 166.34.245.230
Destn IP Port (Media): 20644
Destn SIP Addr (Control) : 166.34.245.230
Destn SIP Port (Control) : 5060
Destination Name : 166.34.245.230
*Mar 8 17:38:38:
Disconnect Cause (CC) : 16
Disconnect Cause (SIP) : 200
Enter the debug ccsip error command to show SIP SPI errors. This command traces all error messages generated from errors encountered by the SIP subsystem.
debug ccsip errorSyntax Description
This command has no arguments or keywords.
Command Modes
EXEC
Command History
12.1(1)T This command was introduced. 12.1.(3)T The output of the command was changed.
Release
Modification
Examples
From one side of the call, the debug output is as follows:
Router1#deb ccsip error SIP Call error tracing is enabled Router1# *Mar 6 14:16:41: CCSIP-SPI-CONTROL: act_idle_call_setup *Mar 6 14:16:41: act_idle_call_setup:Not using Voice Class Codec *Mar 6 14:16:41: act_idle_call_setup: preferred_codec set[0] type :g711ulaw bytes: 160 *Mar 6 14:16:41: REQUEST CONNECTION TO IP:166.34.245.231 PORT:5060 *Mar 6 14:16:41: CCSIP-SPI-CONTROL: act_idle_connection_created *Mar 6 14:16:41: CCSIP-SPI-CONTROL: act_idle_connection_created: Connid(1) created to 166.34.245.231:5060, local_port 55674 *Mar 6 14:16:41: sipSPIAddLocalContact *Mar 6 14:16:41: CCSIP-SPI-CONTROL: sip_stats_method *Mar 6 14:16:41: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.231:5060 *Mar 6 14:16:41: CCSIP-SPI-CONTROL: act_sentinvite_new_message *Mar 6 14:16:41: CCSIP-SPI-CONTROL: sipSPICheckResponse *Mar 6 14:16:41: CCSIP-SPI-CONTROL: sip_stats_status_code *Mar 6 14:16:41: Roundtrip delay 4 milliseconds for method INVITE *Mar 6 14:16:41: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.231:5060 *Mar 6 14:16:41: CCSIP-SPI-CONTROL: act_recdproc_new_message *Mar 6 14:16:41: CCSIP-SPI-CONTROL: sipSPICheckResponse *Mar 6 14:16:41: CCSIP-SPI-CONTROL: sipSPICheckResponse : Updating session description *Mar 6 14:16:41: CCSIP-SPI-CONTROL: sip_stats_status_code *Mar 6 14:16:41: Roundtrip delay 8 milliseconds for method INVITE *Mar 6 14:16:41: HandleSIP1xxRinging: SDP MediaTypes negotiation successful! Negotiated Codec : g711ulaw , bytes :160 Inband Alerting : 0 *Mar 6 14:16:45: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.231:5060 *Mar 6 14:16:45: CCSIP-SPI-CONTROL: act_recdproc_new_message *Mar 6 14:16:45: CCSIP-SPI-CONTROL: sipSPICheckResponse *Mar 6 14:16:45: CCSIP-SPI-CONTROL: sipSPICheckResponse : Updating session description *Mar 6 14:16:45: CCSIP-SPI-CONTROL: sip_stats_status_code *Mar 6 14:16:45: Roundtrip delay 3844 milliseconds for method INVITE *Mar 6 14:16:45: CCSIP-SPI-CONTROL: act_recdproc_new_message: SDP MediaTypes negotiation successful! Negotiated Codec : g711ulaw , bytes :160 *Mar 6 14:16:45: CCSIP-SPI-CONTROL: sipSPIReconnectConnection *Mar 6 14:16:45: CCSIP-SPI-CONTROL: recv_200_OK_for_invite *Mar 6 14:16:45: CCSIP-SPI-CONTROL: sip_stats_method *Mar 6 14:16:45: HandleUdpReconnection: Udp socket connected for fd: 1 with 166.34.245.231:5060 *Mar 6 14:16:45: CCSIP-SPI-CONTROL: ccsip_caps_ind *Mar 6 14:16:45: ccsip_caps_ind: Load DSP with codec (5) g711ulaw, Bytes=160 *Mar 6 14:16:45: ccsip_caps_ind: set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_INBAND_VOICE *Mar 6 14:16:45: CCSIP-SPI-CONTROL: ccsip_caps_ack *Mar 6 14:16:49: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.231:56101 *Mar 6 14:16:49: CCSIP-SPI-CONTROL: act_active_new_message *Mar 6 14:16:49: CCSIP-SPI-CONTROL: sact_active_new_message_request *Mar 6 14:16:49: CCSIP-SPI-CONTROL: sip_stats_method *Mar 6 14:16:49: CCSIP-SPI-CONTROL: sip_stats_status_code *Mar 6 14:16:49: CCSIP-SPI-CONTROL: sipSPIInitiateCallDisconnect : Initiate call disconnect(16) for outgoing call *Mar 6 14:16:49: CCSIP-SPI-CONTROL: act_disconnecting_disconnect *Mar 6 14:16:49: CCSIP-SPI-CONTROL: sipSPICallCleanup *Mar 6 14:16:49: CLOSE CONNECTION TO CONNID:1 *Mar 6 14:16:49: sipSPIIcpifUpdate :CallState: 4 Playout: 2945 DiscTime:48340988 ConnTime 48340525 *Mar 6 14:16:49: udpsock_close_connect: Socket fd: 1 closed for connid 1 with remote port: 5060 Router1#
From the other side of the call, the debug output is as follows:
Router2#debug ccsip error SIP Call error tracing is enabled Router2# *Mar 8 17:42:39: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.230:55674 *Mar 8 17:42:39: CCSIP-SPI-CONTROL: sipSPISipIncomingCall *Mar 8 17:42:39: CCSIP-SPI-CONTROL: act_idle_new_message *Mar 8 17:42:39: CCSIP-SPI-CONTROL: sact_idle_new_message_invite *Mar 8 17:42:39: CCSIP-SPI-CONTROL: sip_stats_method *Mar 8 17:42:39: sact_idle_new_message_invite:Not Using Voice Class Codec *Mar 8 17:42:39: sact_idle_new_message_invite: Preferred codec[0] type: g711ulaw Bytes :160 *Mar 8 17:42:39: sact_idle_new_message_invite: Media Negotiation successful for an incoming call *Mar 8 17:42:39: sact_idle_new_message_invite: Negotiated Codec : g711ulaw, bytes :160 Preferred Codec : g711ulaw, bytes :160 *Mar 8 17:42:39: CCSIP-SPI-CONTROL: sip_stats_status_code *Mar 8 17:42:39: Num of Contact Locations 1 3660110 166.34.245.230 5060 *Mar 8 17:42:39: CCSIP-SPI-CONTROL: act_recdinvite_proceeding *Mar 8 17:42:39: CCSIP-SPI-CONTROL: ccsip_caps_ind *Mar 8 17:42:39: ccsip_caps_ind: codec(negotiated) = 5(Bytes 160) *Mar 8 17:42:39: ccsip_caps_ind: Load DSP with codec (5) g711ulaw, Bytes=160 *Mar 8 17:42:39: ccsip_caps_ind: set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_INBAND_VOICE *Mar 8 17:42:39: CCSIP-SPI-CONTROL: ccsip_caps_ack *Mar 8 17:42:39: CCSIP-SPI-CONTROL: act_recdinvite_alerting *Mar 8 17:42:39: 180 Ringing with SDP - not likely *Mar 8 17:42:39: CCSIP-SPI-CONTROL: sip_stats_status_code *Mar 8 17:42:42: CCSIP-SPI-CONTROL: act_sentalert_connect *Mar 8 17:42:42: sipSPIAddLocalContact *Mar 8 17:42:42: CCSIP-SPI-CONTROL: sip_stats_status_code *Mar 8 17:42:42: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.230:55674 *Mar 8 17:42:42: CCSIP-SPI-CONTROL: act_sentsucc_new_message *Mar 8 17:42:42: CCSIP-SPI-CONTROL: sip_stats_method *Mar 8 17:42:47: CCSIP-SPI-CONTROL: act_active_disconnect *Mar 8 17:42:47: REQUEST CONNECTION TO IP:166.34.245.230 PORT:5060 *Mar 8 17:42:47: CCSIP-SPI-CONTROL: act_active_connection_created *Mar 8 17:42:47: CCSIP-SPI-CONTROL: sipSPICheckSocketConnection *Mar 8 17:42:47: CCSIP-SPI-CONTROL: sipSPICheckSocketConnection: Connid(1) created to 166.34.245.230:5060, local_port 56101 *Mar 8 17:42:47: CCSIP-SPI-CONTROL: sip_stats_method *Mar 8 17:42:47: HandleUdpSocketReads :Msg enqueued for SPI with IPaddr: 166.34.245.230:55674 *Mar 8 17:42:47: CCSIP-SPI-CONTROL: act_disconnecting_new_message *Mar 8 17:42:47: CCSIP-SPI-CONTROL: sact_disconnecting_new_message_response *Mar 8 17:42:47: CCSIP-SPI-CONTROL: sipSPICheckResponse *Mar 8 17:42:47: CCSIP-SPI-CONTROL: sip_stats_status_code *Mar 8 17:42:47: Roundtrip delay 0 milliseconds for method BYE *Mar 8 17:42:47: CCSIP-SPI-CONTROL: sipSPICallCleanup *Mar 8 17:42:47: CLOSE CONNECTION TO CONNID:1 *Mar 8 17:42:47: sipSPIIcpifUpdate :CallState: 4 Playout: 1255 DiscTime:66856757 ConnTime 66856294 *Mar 8 17:42:47: udpsock_close_connect: Socket fd: 1 closed for connid 1 with remote port: 5060
Enter the debug ccsip events command to show all SIP SPI events tracing. This command traces the events posted to SIP SPI from all interfaces.
debug ccsip eventsSyntax Description
This command has no arguments or keywords.
Command Modes
EXEC
Command History
12.1(1)T This command was introduced. 12.1.(3)T The output of the command was changed.
Release
Modification
Examples
From one side of the call, the debug output is as follows:
Router1#debug ccsip events SIP Call events tracing is enabled Router1# *Mar 6 14:17:57: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP *Mar 6 14:17:57: Queued event from SIP SPI : SIPSPI_EV_CREATE_CONNECTION *Mar 6 14:17:57: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE *Mar 6 14:18:00: Queued event from SIP SPI : SIPSPI_EV_RECONNECT_CONNECTION *Mar 6 14:18:00: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE *Mar 6 14:18:04: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE *Mar 6 14:18:04: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_DISCONNECT *Mar 6 14:18:04: Queued event from SIP SPI : SIPSPI_EV_CLOSE_CONNECTION Router1#
From the other side of the call, the debug output is as follows:
Router2#deb ccsip events SIP Call events tracing is enabled Router2# *Mar 8 17:43:55: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE *Mar 8 17:43:55: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_PROCEEDING *Mar 8 17:43:55: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_ALERTING *Mar 8 17:43:55: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE *Mar 8 17:43:58: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_CONNECT *Mar 8 17:43:58: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE *Mar 8 17:44:01: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_DISCONNECT *Mar 8 17:44:01: Queued event from SIP SPI : SIPSPI_EV_CREATE_CONNECTION *Mar 8 17:44:01: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE *Mar 8 17:44:01: Queued event from SIP SPI : SIPSPI_EV_CLOSE_CONNECTION
Enter the debug ccsip messages command to show all SIP SPI message tracing. This command traces the SIP messages exchanged between the SIP UA client (UAC) and the access server.
debug ccsip messagesSyntax Description
This command has no arguments or keywords.
Command Modes
EXEC
Command History
12.1(1)T This command was introduced. 12.1.(3)T The output of the command was changed.
Release
Modification
Examples
From one side of the call, the debug output is as follows:
Router1#debug ccsip message SIP Call messages tracing is enabled Router1# *Mar 6 14:19:14: Sent: INVITE sip:3660210@166.34.245.231;user=phone;phone-context=unknown SIP/2.0 Via: SIP/2.0/UDP 166.34.245.230:55820 From: "3660110" <sip:3660110@166.34.245.230> To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown> Date: Sat, 06 Mar 1993 19:19:14 GMT Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194 Cisco-Guid: 2881152943-2184249568-0-483551624 User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 101 INVITE Max-Forwards: 6 Timestamp: 731427554 Contact: <sip:3660110@166.34.245.230:5060;user=phone> Expires: 180 Content-Type: application/sdp Content-Length: 138 v=0 o=CiscoSystemsSIP-GW-UserAgent 5596 7982 IN IP4 166.34.245.230 s=SIP Call t=0 0 c=IN IP4 166.34.245.230 m=audio 20762 RTP/AVP 0 *Mar 6 14:19:14: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 166.34.245.230:55820 From: "3660110" <sip:3660110@166.34.245.230> To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown> Date: Mon, 08 Mar 1993 22:45:12 GMT Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194 Timestamp: 731427554 Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 101 INVITE Content-Length: 0 *Mar 6 14:19:14: Received: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 166.34.245.230:55820 From: "3660110" <sip:3660110@166.34.245.230> To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown> Date: Mon, 08 Mar 1993 22:45:12 GMT Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194 Timestamp: 731427554 Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 101 INVITE Content-Type: application/sdp Content-Length: 138 v=0 o=CiscoSystemsSIP-GW-UserAgent 1193 7927 IN IP4 166.34.245.231 s=SIP Call t=0 0 c=IN IP4 166.34.245.231 m=audio 20224 RTP/AVP 0 *Mar 6 14:19:16: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 166.34.245.230:55820 From: "3660110" <sip:3660110@166.34.245.230> To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27DBC6D8-1357 Date: Mon, 08 Mar 1993 22:45:12 GMT Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194 Timestamp: 731427554 Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled Contact: <sip:3660210@166.34.245.231:5060;user=phone> CSeq: 101 INVITE Content-Type: application/sdp Content-Length: 138 v=0 o=CiscoSystemsSIP-GW-UserAgent 1193 7927 IN IP4 166.34.245.231 s=SIP Call t=0 0 c=IN IP4 166.34.245.231 m=audio 20224 RTP/AVP 0 *Mar 6 14:19:16: Sent: ACK sip:3660210@166.34.245.231:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 166.34.245.230:55820 From: "3660110" <sip:3660110@166.34.245.230> To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27DBC6D8-1357 Date: Sat, 06 Mar 1993 19:19:14 GMT Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194 Max-Forwards: 6 Content-Type: application/sdp Content-Length: 138 CSeq: 101 ACK v=0 o=CiscoSystemsSIP-GW-UserAgent 5596 7982 IN IP4 166.34.245.230 s=SIP Call t=0 0 c=IN IP4 166.34.245.230 m=audio 20762 RTP/AVP 0 *Mar 6 14:19:19: Received: BYE sip:3660110@166.34.245.230:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 166.34.245.231:53600 From: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27DBC6D8-1357 To: "3660110" <sip:3660110@166.34.245.230> Date: Mon, 08 Mar 1993 22:45:14 GMT Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194 User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled Max-Forwards: 6 Timestamp: 731612717 CSeq: 101 BYE Content-Length: 0 *Mar 6 14:19:19: Sent: SIP/2.0 200 OK Via: SIP/2.0/UDP 166.34.245.231:53600 From: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27DBC6D8-1357 To: "3660110" <sip:3660110@166.34.245.230> Date: Sat, 06 Mar 1993 19:19:19 GMT Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194 Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled Timestamp: 731612717 Content-Length: 0 CSeq: 101 BYE Router1#
From the other side of the call, the debug output is as follows:
Router2#debug ccsip message SIP Call messages tracing is enabled Router2# *Mar 8 17:45:12: Received: INVITE sip:3660210@166.34.245.231;user=phone;phone-context=unknown SIP/2.0 Via: SIP/2.0/UDP 166.34.245.230:55820 From: "3660110" <sip:3660110@166.34.245.230> To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown> Date: Sat, 06 Mar 1993 19:19:14 GMT Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194 Cisco-Guid: 2881152943-2184249568-0-483551624 User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 101 INVITE Max-Forwards: 6 Timestamp: 731427554 Contact: <sip:3660110@166.34.245.230:5060;user=phone> Expires: 180 Content-Type: application/sdp Content-Length: 138 v=0 o=CiscoSystemsSIP-GW-UserAgent 5596 7982 IN IP4 166.34.245.230 s=SIP Call t=0 0 c=IN IP4 166.34.245.230 m=audio 20762 RTP/AVP 0 *Mar 8 17:45:12: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 166.34.245.230:55820 From: "3660110" <sip:3660110@166.34.245.230> To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown> Date: Mon, 08 Mar 1993 22:45:12 GMT Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194 Timestamp: 731427554 Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 101 INVITE Content-Length: 0 *Mar 8 17:45:12: Sent: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 166.34.245.230:55820 From: "3660110" <sip:3660110@166.34.245.230> To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown> Date: Mon, 08 Mar 1993 22:45:12 GMT Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194 Timestamp: 731427554 Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 101 INVITE Content-Type: application/sdp Content-Length: 138 v=0 o=CiscoSystemsSIP-GW-UserAgent 1193 7927 IN IP4 166.34.245.231 s=SIP Call t=0 0 c=IN IP4 166.34.245.231 m=audio 20224 RTP/AVP 0 *Mar 8 17:45:14: Sent: SIP/2.0 200 OK Via: SIP/2.0/UDP 166.34.245.230:55820 From: "3660110" <sip:3660110@166.34.245.230> To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27DBC6D8-1357 Date: Mon, 08 Mar 1993 22:45:12 GMT Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194 Timestamp: 731427554 Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled Contact: <sip:3660210@166.34.245.231:5060;user=phone> CSeq: 101 INVITE Content-Type: application/sdp Content-Length: 138 v=0 o=CiscoSystemsSIP-GW-UserAgent 1193 7927 IN IP4 166.34.245.231 s=SIP Call t=0 0 c=IN IP4 166.34.245.231 m=audio 20224 RTP/AVP 0 *Mar 8 17:45:14: Received: ACK sip:3660210@166.34.245.231:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 166.34.245.230:55820 From: "3660110" <sip:3660110@166.34.245.230> To: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27DBC6D8-1357 Date: Sat, 06 Mar 1993 19:19:14 GMT Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194 Max-Forwards: 6 Content-Type: application/sdp Content-Length: 138 CSeq: 101 ACK v=0 o=CiscoSystemsSIP-GW-UserAgent 5596 7982 IN IP4 166.34.245.230 s=SIP Call t=0 0 c=IN IP4 166.34.245.230 m=audio 20762 RTP/AVP 0 *Mar 8 17:45:17: Sent: BYE sip:3660110@166.34.245.230:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 166.34.245.231:53600 From: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27DBC6D8-1357 To: "3660110" <sip:3660110@166.34.245.230> Date: Mon, 08 Mar 1993 22:45:14 GMT Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194 User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled Max-Forwards: 6 Timestamp: 731612717 CSeq: 101 BYE Content-Length: 0 *Mar 8 17:45:17: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 166.34.245.231:53600 From: <sip:3660210@166.34.245.231;user=phone;phone-context=unknown>;tag=27DBC6D8-1357 To: "3660110" <sip:3660110@166.34.245.230> Date: Sat, 06 Mar 1993 19:19:19 GMT Call-ID: ABBAE7AF-823100E2-0-1CD274BC@172.18.192.194 Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled Timestamp: 731612717 Content-Length: 0 CSeq: 101 BYE
Enter the debug ccsip states command to show all SIP SPI state tracing. This command traces the state machine changes of SIP SPI and displays the state transitions.
debug ccsip statesSyntax Description
This command has no arguments or keywords.
Command Modes
EXEC
Command History
12.1(1)T This command was introduced.
Release
Modification
Examples
Router1#deb ccsip states SIP Call states tracing is enabled Router1# *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
Table 1 lists the PSTN cause codes that can be sent as an ISDN cause information element (IE) and the corresponding SIP event for each.
PSTN Cause Code | Description | SIP Event |
|---|---|---|
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| 480 Temporarily unavailable |
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| 503 Service unavailable |
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| 501 Not implemented |
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| 501 Not implemented |
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| 400 Bad request |
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Any code other than those listed above |
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Table 2 lists the SIP events and the corresponding PSTN cause codes for each.
SIP Event | PSTN Cause Code | Description |
|---|---|---|
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| 127 | Interworking |
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| 127 | Interworking |
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| 127 | Interworking |
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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.
call—In SIP, a call consists of all participants in a conference invited by a common source. A SIP call is identified by a globally unique call ID. A point-to-point IP telephony conversation maps into a single SIP call. For a multicast session, each participant in the session constitutes a unique call. Each call involves a UAC and a UAS application.
CAS—Channel associated signaling.
CCAPI—Call control applications programming interface.
CLI—Command line interface.
CO—Central office.
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 (V0IP), there are two types of dial peers: POTS and VoIP.
DNS—Domain name system used to address translation to convert H.323 IDs, URLs, or e-mail IDs to IP addresses. DNS is also used to assist in the locating remote gatekeepers and to reverse-map raw IP addresses to host names of administrative domains.
DNIS—Dialed number identification service (the called number).
DSP—Digital signal processor.
DTMF—Dual tone multi-frequency.
E.164—The international public telecommunications numbering plan. A standard set by ITU-T which addresses telephone numbers.
E&M—Ear and mouth RBS signaling.
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.
IPSEC—An IETF standard that is used to provide security for transmission of sensitive information over unprotected networks such as the Internet. IPSec acts at the network layer, protecting and authenticating IP packets between participating IPSec devices ("peers"), such as Cisco routers.
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.
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—Multi-frequency 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—A 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.
PSTN—Public switched telephone network. PSTN refers to the local telephone company.
QoS—Quality of Service. Measure of performance for a transmission system that reflects its transmission quality and service availability. QoS refers to the ability of a network to provide better service to selected network traffic over various underlying technologies. QoS is not inherent in a network infrastructure. Rather, you must institute QoS by strategically deploying features that implement it throughout the network.
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.
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.
session—In SIP, a session is a set of multimedia senders and receivers and the data streams flowing between the senders and receivers. A SIP multimedia conference is an example of a session. A callee can be invited several times, by different calls, to the same session.
SIP—Session Initiation Protocol. This is an application-layer protocol developed by the IETF MMUSIC Working Group to equip platforms to signal the setup of voice and multimedia calls over IP networks. SIP features are compliant with IETF RFC 2543, published in March 1999.
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 and UAC.
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: Tue Sep 19 17:28:23 PDT 2000
Copyright 1989-2000©Cisco Systems Inc.