The show commands listed in Table 5-2 are valuable when examining the status of SIP
components and troubleshooting SIP environments.
SIP show Commands
Command Description
show sip service Displays the status of the SIP VoIP service
show sip-ua status Displays the status of the SIP UA
show sip-ua register status Displays the status of E.164 numbers that a SIP gateway has
registered with an external primary SIP registrar
show sip-ua timers Displays SIP UA timers
show sip-ua connections Displays active SIP UA connections
show sip-ua calls Displays active SIP UA calls
show sip-ua statistics Displays SIP traffic statistics
Use the show sip service command to display the status of SIP call service on a SIP gateway.
Example 5-16 provides sample output from the show sip service command.
Example 5-16 show sip service Command
310 Authorized Self-Study Guide: Cisco Voice over IP (CVOICE)
Router#show sip service
SIP Service is up
Use the show sip-ua status command to display the status for the SIP user agent, including
whether call redirection is enabled or disabled. Example 5-17 provides sample output
from the show sip-ua status command.
Example 5-17 show sip-ua status Command
Router#show sip-ua status
SIP User Agent Status
SIP User Agent for UDP : ENABLED
SIP User Agent for TCP : ENABLED
SIP User Agent bind status(signaling): DISABLED
SIP User Agent bind status(media): DISABLED
SIP max-forwards : 6
SIP DNS SRV version: 1 (rfc 2052)
Redirection (3xx) message handling: ENABLED
Use the show sip-ua timers command to display the current settings for the SIP useragent
timers. Example 5-18 provides sample output from the show sip-ua timers
command.
Example 5-18 show sip-ua timers Command
Chapter 5: Examining VoIP Gateways and Gateway Control Protocols 311
Router#show sip-ua timers
SIP UA Timer Values (millisecs)
trying 500, expires 180000, connect 500, disconnect 500
comet 500, prack 500, rel1xx 500, notify 500
refer 500, register 500
Use the show sip-ua register status command to display the status of E.164 numbers
that a SIP gateway has registered with an external primary SIP registrar. Example 5-19
provides sample output from the show sip-ua register status command.
Example 5-19 show sip-ua register status Command
Router#show sip-ua register status
Line peer expires(sec) registered
4001 20001 596 no
4002 20002 596 no
5100 1 596 no
9998 2 596 no
Example 5-20 shows the output of the show sip-ua calls command, which provides
detailed information about current SIP calls.
Example 5-20 show sip-ua calls Command
router#show sip-ua calls
SIP UAC CALL INFO
Number of SIP User Agent Client(UAC) calls: 0
SIP UAS CALL INFO
Call 1
SIP Call ID : D215F304-7B5A11DC-8005EA1A-6A8F4AD@10.10.10.2
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 2818902001
Called Number : 1003
Bit Flags : 0x1212003A 0x100000 0x488
CC Call ID : 1
Source IP Address (Sig ): 10.10.10.1
Destn SIP Req Addr:Port : 10.10.10.2:5060
Destn SIP Resp Addr:Port: 10.10.10.2:56884
Destination Name : 10.10.10.2
Example 5-20 show sip-ua calls Command continued
312 Authorized Self-Study Guide: Cisco Voice over IP (CVOICE)
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 1
Stream Type : voice-only (0)
Negotiated Codec : g729r8 (20 bytes)
Codec Payload Type : 18
Negotiated Dtmf-relay : inband-voice
Dtmf-relay Payload Type : 0
Media Source IP Addr:Port: 10.10.10.1:18050
Media Dest IP Addr:Port : 10.10.10.2:16522
Orig Media Dest IP Addr:Port : 0.0.0.0:0
Number of SIP User Agent Server(UAS) calls: 1
The following debug commands are valuable when examining the status of SIP components
and troubleshooting SIP environments:
■ debug asnl events: Use this command to verify that the SIP subscription server is
up. The output displays a pending message if, for example, the client is unsuccessful
in communicating with the server.
■ debug voip ccapi inout: This command shows every interaction with the call control
API on both the telephone interface and on the VoIP side. By monitoring the output,
you can follow the progress of a call from the inbound interface or VoIP peer to the
outbound side of the call. This debug command is very active. Therefore, you must
use it sparingly in a live network.
■ debug voip ccapi protoheaders: This command displays messages sent between the
originating and the terminating gateways. If no headers are being received by the terminating
gateway, verify that the header-passing command is enabled on the originating
gateway.
■ debug ccsip all: This command enables all ccsip-type debugging. This debug command
is very active. Therefore, you should use it sparingly in a live network.
■ debug ccsip calls: This command displays all SIP call details as they are updated in
the SIP call control block. You can use this debug command to monitor call records
for suspicious clearing causes.
■ debug ccsip errors: This command traces all errors encountered by the SIP
subsystem.
■ debug ccsip events: This command traces events, such as call setups, connections,
and disconnections. An events version of a debug command is often the best place
to start because detailed debugs provide a great deal of useful information.
■ debug ccsip info: This command enables tracing of general SIP Service Provider
Interface (SPI) information, including verification that call redirection is disabled.
■ debug ccsip media: This command enables tracing of SIP media streams.
■ debug ccsip messages: This command shows the headers of SIP messages that are
exchanged between a client and a server.
■ debug ccsip preauth: This command enables diagnostic reporting of authentication,
authorization, and accounting (AAA) for SIP calls.
■ debug ccsip states: This command displays the SIP states and state changes for sessions
within the SIP subsystem.
■ debug ccsip transport: This command enables tracing of the SIP transport handler
and the TCP or UDP process.
Examples 5-21, 5-22, and 5-23 show what a successful SIP session between two endpoints
looks like in the output of the debug ccsip messages command. Example 5-21
shows a SIP INVITE message being sent from one phone to another.
Example 5-21 INVITE Message
Chapter 5: Examining VoIP Gateways and Gateway Control Protocols 313
HQ-1#debug ccsip messages
SIP Call messages tracing is enabled
HQ-1#
*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”
To:
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:
Expires: 180
Content-Type: application/sdp
Content-Length: 138
Example 5-22 shows the other endpoint returning an OK. Notice the Contact information
added to the output.
Example 5-22 OK Message
314 Authorized Self-Study Guide: Cisco Voice over IP (CVOICE)
*Mar 6 14:19:16: Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 166.34.245.230:55820
From: “3660110”
To:
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:
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
Example 5-23 shows the other endpoint ending the session with a BYE message.
Example 5-23 BYE Message
*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:
1357
To: “3660110”
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
The capital capacity covered in this affiliate are the following:
■ ITU-T Recommendation H.323 describes an basement of terminals, common
control components, services, and protocols that are acclimated for multimedia
communications.
■ Functional apparatus of H.323 accommodate terminals, gateways, gatekeepers, Cisco
UBEs, and MCUs.
■ Calls can be accustomed amid endpoints, endpoints to gatekeepers, or gatekeepers
to gatekeepers.
■ H.323 calls can action with or after the use of a gatekeeper.
■ H.323 defines three types of multipoint conferences.
■ When configuring codecs, you can specify one codec or set up codec negotiation.
■ You ability appetite to acclimatize some of the H.323 timers to accommodated arrangement requirements.
■ You can use several commands to configure fax appearance on H.323 gateways.
■ DTMF broadcast solves the botheration of DTMF distortion.
■ Use the appearance aperture command to verify H.323 aperture status.
■ MGCP defines an ambiance for authoritative telephony gateways from a centralized
call agent.
■ MGCP apparatus accommodate endpoints, gateways, and
call agents.
■ Calls and access are basal concepts in MGCP.
■ MGCP alarm breeze consists of an barter of letters amid a alarm abettor and a
gateway.
■ The mgcp command can be acclimated to configure residential and block gateways on a
Cisco router.
■ Several appearance and alter commands advice to verify an MGCP configuration.
■ SIP is authentic by IETF RFCs 2543 and 3261 and allows affiliation with third-party
VoIP networks.
■ SIP is modeled on the interworking of UAs and arrangement servers.
■ A SIP alarm breeze consists of signaling and manual of agent and media packets.
■ Communication amid SIP apparatus uses a appeal and acknowledgment message
model.
■ A SIP abode consists of an alternative user ID, a host description, and optional
parameters to authorize the abode added precisely.
■ SIP alarm bureaucracy models accommodate direct, proxy server, and redirection.
■ You can use several commands on Cisco IOS to configure SIP on Cisco IOS routers.
■ You can use several commands on Cisco IOS to verify and troubleshoot a SIP
integration.