Table of Contents
Managing and Troubleshooting the Cisco VoIP Infrastructure Solution for SIP
Using CVM 2.0 to Manage the Cisco VoIP Infrastructure Solution for SIP
Troubleshooting the Cisco VoIP Infrastructure Solution for SIP
Managing and Troubleshooting the Cisco VoIP Infrastructure Solution for SIP
This chapter describes tools that you can use to manage and troubleshoot the Cisco VoIP Infrastructure Solution for SIP. It also includes tips for problem isolation and suggested actions for resolution. It includes the following sections:
Using CVM 2.0 to Manage the Cisco VoIP Infrastructure Solution for SIP
Ciscoworks2000 Voice Manager (CVM) is a client-server, web-based voice management solution used by network administrators to configure and manage voice ports and create and modify dial plans on voice-enabled Cisco routers. Using CVM, network administrators can:
- Manage the configuration of FXO, FXS, E&M, and ISDN voice interfaces on voice-enabled routers
- Create and manage local (POTS) dial plans on voice-enabled routers
- Generate detailed reports using Telemate.net Quickview
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Note CVM is not a device configuration tool. Devices supported by CVM must first be configured through
the CLI and have Simple Network Management Protocol (SNMP) enabled before they can be
managed by CVM. You can then use CVM to modify the configuration of voice ports and create and
manage local and network dial plans.
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Note For complete information about using CVM to manage your SIP VoIP infrastructure, see the
CiscoWorks2000 Voice Manager 2.0 documentation.
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Prerequisites
The CVM Server requires the following:
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Note System requirements for the server are based on software requirements and a call volume of
96,000 calls per hour. The call volume is based on an estimated 20 calls per DS0 channel, 3
minutes holding time, and 60 busy minutes.
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- 256 MB of memory
- CPU running at 450 MHz
- 8 GB of available hard disk space
- Windows NT 4.0 with Service Pack 5
- CiscoWorks2000 CD One
The CVM Client requires the following:
- 64 MB of memory
- CPU running at 300 MHz
- One of the following:
- Windows 95 running Netscape 4.04 or Internet Explorer 4.01 and 64 MB of virtual memory
- Windows NT running Netscape 4.04 or Internet Explorer 4.01 and 64 MB of virtual memory
- Solaris running Netscape 4.04 with Telnet and Java enabled and 64 MB of virtual memory
- Display settings of:
- 1024 x 768 resolution
- 16-bit color palette
Before you can use CVM to manage your voice network, for each router that you are going to add to CVM:
- You must have network access to the router.
- You must know the IP address of the router.
- You must know all the passwords for the router.
- You must know the SNMP read community string for the router.
- Telnet must be enabled on the router. Because Telnet is used by CVM to communicate with a router, session timeout should be configured to a non-zero value for all tty lines.
- SNMP must be enabled on the router.
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Note For detailed information about using CVM to manage your SIP VoIP infrastructure, see the
CiscoWorks2000 Voice Manager 2.0 documentation.
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Troubleshooting the Cisco VoIP Infrastructure Solution for SIP
This section provides procedural and reference information that you can use to determine and resolve problems you might experience while using the SIP components of the Cisco VoIP Infrastructure Solution for SIP.
This section contains the following information:
Troubleshooting the Cisco SIP IP Phone 7960
This section describes troubleshooting features and tips for the Cisco SIP IP phone 7960.
Troubleshooting Features
The following is a list of features on the Cisco SIP IP phone that you can use to troubleshoot phone:
- Settings button to Network Configuration soft key—Use to view or modify the network configuration of the phone.
- Settings button to SIP Configuration soft key—Use to view or modify a phone's SIP settings.
- Settings button to Status—Display configuration or initialization errors.
- Call Messages on LED screen—Display basic SIP message flows.
- Pressing "i" key twice during a call—Displays real-time transferring and receiving call statistics. This option is recommended for trouble-shooting voice-quality issues.
In addition to the features listed above, the RS-232 port located on the back of the Cisco SIP IP phone 7960 is a console port and can be used to gather debug information.
The RS-232 port is password-protected and requires a custom RJ-11-to-RJ-45 cable.
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Note For a PC connection, the RJ-45 connection needs a DB-9 female DTE adapter or an RJ-45 crossover
cable for an octal async connection. The password "cisco" must be entered to enable any output to
be seen via the RS-232 port. The connection baud rate, parity, start bits, and stop bits are 9600, N, 8,
and 1.
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To use the console port, use a RJ-11-to-RJ-45 custom cable to connect the RS-232 port to a PC.
Table 5-1 lists the RJ-11-to-RJ-45 cable pinouts.
Table 5-1 Pinouts
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RJ-11 or RJ-12 |
RJ-45 |
2
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6
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3
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4
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4
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3
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To connect the console port, complete the following tasks:
Step 1 Insert the RJ-11 end of the rolled cable into the RS-232 port on the back of the phone.
Step 2 Use an RJ-45-to-DB-9 female DTE adapter (labeled "TERMINAL") to connect the console port to a PC running terminal emulation software.
Step 3 Insert the RJ-45 end of the rollover cable into the DTE adapter.
Step 4 From the console terminal, start the terminal emulation program.
Step 5 Type "cisco". A prompt is displayed.
At the prompt, you can issue the following commands to assist you in troubleshooting and debugging the phone:
- debug error—Displays error messages that are occurring in the call flow process.
- debug sip-message—Enables you to view a text display of a call flow.
Troubleshooting Tips
This section provides tips for resolving the following Cisco SIP IP phone problems:
Phone is Unprovisioned or is Unable to Obtain an IP Address
To determine why a phone is unprovisioned or unable to obtain an IP address, perform the following tasks as necessary:
- If using TFTP to download configuration files, verify that the SIPDefault.cnf and the phone-specific configuration (SIPmac.cnf where mac is the MAC address of the phone) files exist and are configured correctly.
- Verify that the TFTP server is working properly.
- Verify that the Cisco SIP IP phone Network Configuration parameters are properly configured and the phone is obtaining the proper IP addressing information (IP address, subnet mask, default gateway, TFTP server, etc.).
- Press the settings button, select Status, and then Status Messages to view messages for missing files or other errors.
- If the DHCP server is on a different IP subnet mask than the Cisco SIP IP phone, verify that "ip helper-address" address is enabled on the local router.
- Verify that the Cisco SIP IP phone software image (P0S3xxyy.bin where xx is the version number and yy is the subversion number) was downloaded from CCO using binary format.
Cisco SIP IP Phone will not Register with the SIP Proxy/Registrar Server
To determine why a phone will not register with a SIP proxy/registrar server, perform the following tasks as necessary:
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Note The character "x" displayed to the right of a line icon indicates that registration has failed.
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- Verify that phone registration with a proxy server is enabled (via the proxy_register parameter in the configuration files). By default, registration during initialization is disabled.
- Verify that the IP address (proxy1_address parameter) of the primary SIP proxy server to be used by the phones is valid.
- If a Fully Qualified Domain Name (FQDN) is specified in the proxy1_address parameter, verify that the DNS server is configured to resolve the FQDN as a DNS A-record type.
- Verify whether the Cisco SIP Proxy Server has been configured to require authentication. If so, ensure that an authentication name and password has been defined in the Cisco SIP IP phone phone-specific configuration file (via the linex_authname and linex_password parameters).
- The Cisco SIP IP phone currently supports the HTTP Digest authentication method. Verify that the authentication method required by the Cisco SIP Proxy Server (via the AuthScheme directive in the sipd.conf file) is HTTP Digest.
- Verify that a registration request hasn't expired. By default, Cisco SIP IP phones will re-register every 3600 sections but this value can be modified via the time_reqister_expires parameter.
Outbound Calls Cannot be Placed from a Cisco SIP IP Phone
If a call cannot be placed from a Cisco SIP IP phone, perform the following tasks as necessary:
- Verify that the Cisco SIP IP phone Network Configuration IP address parameters have been correctly entered or received from a DHCP server.
- Verify that the Cisco SIP Proxy Server used by the phone is working properly.
- Verify that the remote SIP device is available.
- Verify that a dial plan has been defined via the dialplan.xml file and if so, that the configuration is correct. This file should have been downloaded from CCO to the root directory of your TFTP server.
- Determine the error tones that are being received and map those tones to the messages displayed on the phone's LCD (SIP 4xx messages, etc.)
- Verify that the Cisco SIP Proxy Server is correctly configured for routes or registrations to the remote destination.
Inbound Calls Cannot be Received on a Cisco SIP IP Phone
If inbound calls cannot be received on a Cisco SIP IP phone, perform the following tasks as necessary:
- Verify that the line (user portion) was defined in the Request-URI or the SIP INVITE request. The Cisco SIP IP phone requires this information to determine the proper line to ring.
- Verify that the Request-URI is sent to port 5060 of the phone's IP address. The phone listens on UDP port 5060.
- Verify that the Cisco SIP IP phone is registered with the local proxy server.
Poor Voice Quality on the Cisco SIP IP Phone
If a call's voice quality is compromised on the Cisco SIP IP phone, perform the following tasks as necessary:
- Check the network path for errors, packet drops, loss, loops, etc.
- Verify that the ToS level for the media stream being used have been correctly set (via the tos_media parameter in the configuration file).
- Verify that the Cisco SIP IP phone is plugged into a switch rather than a hub to avoid excessive collisions and packet loss.
- Ensure that there is enough bandwidth on the network for the selected codec (especially for calls over a WAN).
- Press the blue "i" button twice on the phone during the call to view realtime transferring and receiving call statistics
- Determine whether the problem just occurs with the handset, headset, or speaker phone, or with all of them.
DTMF Digits Do Not Function Properly
If DTMF digits are not functioning properly, perform the following tasks as necessary:
- If out-of-bound signaling via the AVT tone method has been enabled (via the dtmf_outofband configuration file parameter), verify that the remote device supports AVT tones (as defined in RFC 2833). If AVT tones have been enabled and the remote device does not support AVT tone, check for packet loss in the end-to-end path.
- Verify which codec is being used. Lower bandwidth codecs yield poorer result if AVT tones are not support because the DTMF digits are carried via audio.
- Verify the length of the tones being created. The tone must be a minimum signal duration of 40 ms with signaling velocity (tone and pause) of no less than 93 ms (as defined in RFC 2833).
Cisco SIP IP Phones do not Work When Plugged into a Line-Powered Switch
If the Cisco SIP IP phones do not work when plugged into a line-powered switch, perform the following task:
- Verify that the phone is running Version 2.0 of the Cisco SIP IP Phone software. (Line-powered support was not available in Version 1.0).
- Verify that the network media type Network Settings parameter is set to auto-negotiation (auto).
Call Transfer Does Not Work Correctly
If call transfer does not work, verify the remote SIP device that is sending the call is using the SIP BYE/Also: method (as defined in Internet draft sip-cc-01.txt.
Some SIP Messages are Retransmitted Too Often
The Cisco SIP IP phone has several timers (INVITE request retries, BYE request retries, etc.) that can be configured via the sip_invite_retx and sip_retx configuration file parameters. In most networks, the default values work fine, however, conditions such as network delay, slower-processing proxy servers, and packet loss might require that the timers be adjusted. If some SIP messages appear to be retransmitted too often, adjust these parameters.
Troubleshooting the Cisco SIP Gateway
This section describes troubleshooting features and tips for Cisco SIP Gateways running Cisco IOS Release 12 1(5)XM.
Troubleshooting Features
The following commands can be used to troubleshoot the Cisco SIP Gateway:
- show sip ?—Displays the different show sip commands.
retry Display SIP Protocol Retry Counts
statistics Display SIP UA Statistics
status Display SIP UA Listener Status
timers Display SIP Protocol Timers
- show sip status—Displays the SIP user agent listener status.
sip-2600a#show sip status
SIP User Agent for UDP : ENABLED
SIP User Agent for TCP : ENABLED
- show sip statistics—Displays SIP user agent statistics
router#show sip statistics
SIP Response Statistics (Inbound/Outbound)
Forwarded 0/0, Queued 0/0,
OkCancel 0/0, OkOptions 0/0
Redirection (Inbound only):
MultipleChoice 0, MovedPermanently 0,
MovedTemporarily 0, SeeOther 0,
UseProxy 0, AlternateService 0
BadRequest 0/3, 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
InternalError 0/0, NotImplemented 0/0,
BadGateway 0/0, ServiceUnavail 0/0,
GatewayTimeout 0/0, BadSipVer 0/0
BusyEverywhere 0/0, Decline 0/0,
NotExistAnywhere 0/0, NotAcceptable 0/0
SIP Total Traffic Statistics (Inbound/Outbound)
Invite 3/7, Ack 2/1, Bye 0/2,
Invite 2, Bye 0, Cancel 0, Response 1
- debug sip ?—Displays the different debug ccsip commands.
all Enable all SIP debugging traces
calls Enable CCSIP SPI calls debugging trace
error Enable SIP error debugging trace
events Enable SIP events debugging trace
messages Enable CCSIP SPI messages debugging trace
states Enable CCSIP SPI states debugging trace
From one side of a call, the following is a sample of debug output:
All SIP call tracing enabled
*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)
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
Contact: <sip:3660110@166.34.245.230:5060;user=phone>
Content-Type: application/sdp
o=CiscoSystemsSIP-GW-UserAgent 1212 283 IN IP4 166.34.245.230
*Mar 6 14:10:42: Received:
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
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
*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:
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
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Content-Type: application/sdp
o=CiscoSystemsSIP-GW-UserAgent 969 7889 IN IP4 166.34.245.231
*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
*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:
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
Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
Contact: <sip:3660210@166.34.245.231:5060;user=phone>
Content-Type: application/sdp
o=CiscoSystemsSIP-GW-UserAgent 969 7889 IN IP4 166.34.245.231
*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
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
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
Content-Type: application/sdp
o=CiscoSystemsSIP-GW-UserAgent 1212 283 IN IP4 166.34.245.230
*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
