SIP

Marcelo Zanata
Components

Error Codes

UA (User Agent) any endpoint.

UAC (User Agent Client) UA that initialize the call
UAS (User Agent Server) UA that receive the call
Proxy Server Do call routing, authentication, authorization, address resolution, loop detection. This can
stay int he signaling path or not.
Redirect Server UA and Proxy can contact it and get the response with one or more address for the user.
Cisco Router can act as it.
Registrar Server Keeps track of current location of UA. IOS and CCM can do it.
Location Server maintains the location database of UA
B2BUA (Back-to-back User Agent) a server acting as UAS and UAC at the same-time, re-initializing the
call. CCM can be SIP B2BUA.
Presence Server gather presence form Presentities and subscribe information from Watchers

Class of Response Code Explanation

Informational/
100 Trying
provisional
180 Ringing
181 Call is being forwarded
182 Queued
183 Session Progress
Success
200 OK
Redirection
300 Multiple Choices
301 Moved Permanently
302 Moved Temporarily
305 Use Proxy
380 Alternative Service
Client-Error
400 Bad Request
Methods
401 Unauthorized
Cisco gateways can send and receive:
402 Payment Required
REGISTER: A UA client sends this message to inform a SIP server of its location.
403 Forbidden
INVITE: A caller sends this message to request that another endpoint join a SIP session, such as a
404 Not Found
conference or a call. This message can also be sent during a call to change session parameters.
405 Method Not Allowed
ACK: A SIP UA can receive several responses to an INVITE. This method acknowledges the final response to
406 Not Acceptable
the INVITE.
407 Proxy Auth Required
CANCEL: This message ends a call that has not yet been fully established.
408 Request Timeout
OPTIONS: This message queries the capabilities of a server. Cisco gateways receive these methods only.
410 Gone
BYE: This message ends a session or declines to take a call.
413 Request Entity Too Large
Cisco gateway do not generate:
414 Requested URL Too Large
INFO: This message is used when data is carried within the message body.
415 Unsupported Media Type
PRACK: This message acknowledges receipt of a provisional, or informational, response to a request.
416 Unsupported URI Scheme
REFER This message points to another address to initiate a transfer.
420 Bad Extension
SUBSCRIBE This message lets the server know that you want to be notified if a specific event happens.
421 Extension Required
NOTIFY This message lets the subscriber know that a specified event has occurred. It can also transmit dual
423 Interval Too Brief
tone multifrequency (DTMF) tones.
480 Temporarily Not Available
UPDATE A UAC uses this to change the session parameters, such as codec used or quality of service (QoS)
481 Transaction Does Not Exist
settings, before answering the initial INVITE.
482 Loop Detected
SDP fields
483 Too Many Hops
v: Tells the SDP version
484 Address Incomplete
o: Lists the organization of the calling party
485 Ambiguous
s: Describes the SDP message
486 Busy Here
c: Lists the IP address of the originator
487 Request Terminated
t: Tells the timer value
488 Not Acceptable Here
m: Describes the media that the originator expects
491 Request Pending
a: Gives the media attributes
493 Undecipherable
Server-error
500 Internal Server Error
DTMF Relay
501 Not Implemented
Named Telephony Events (RFC2833) RTP Packets with a different type field (In-band)
502 Bad Gateway
Key Press Markup Language (KPML) SIP Subscriber messages with DTMF in XML like format (OOB)
503 Service Unavailable
Unsolicited Notify (UN) SIP Notify messages and without SIP Subscribe (OOB)
504 Server Timeout
Cisco RTP RTP Packets with a different type field.
505 SIP Version Not Supported
Call flow with multiple servers
513 Message Too Large
Global failure
600 Busy Everywhere
603 Decline
604 Does Not Exist Anywhere
606 Not Acceptable

Dialpeer configuration
dial-peer voice 3401 voip
session target ipv4:10.6.2.1
session protocol sipv2
session transport tcp
!
dial-peer voice 4404 voip
session target sip-server
session protocol sipv2
voice-class sip transpor switch udp tcp
destination-pattern 4404...

voice-class sip transport switch udp tcp switch from

UDP to TCP when a packet gets within 200 bytes of
the MTU to avoid UDP fragmentation.

Other details

SIP UA commands
sip-ua
registrar ipv4:10.30.25.250 tcp
registrar ipv4:10.30.25.251 tcp secon
sip-server ipv4:10.30.25.252
max-forwards 10
no transport udp

Default Ports: 5060 TCP/UDP / TLS: 5061

Plain-Text messages
Sip address is called URI = uniform resource identifier
SIP Dialplan considerations
The default behavior of SIP Phone is compare digits to the internal dial plan. When have a match, its sends
SIP Voice Service commands
an INVITE.
voice service voip
When you use KPML (Key Press Markup Language), the SIP phone sends each digit to CCM that can instruct redirect ip2ip
the phone what do or route the call.
sip

bind control source-interface lo0

registrar server exp max 1500 min 500

SIP

Marcelo Zanata
Early Offer

Delayed Offer

Call flow between two gateways

PBX

GWA

GWB

Early Media

Call Flow using a Proxy Server

Endpoint
Setup

PBX

Setup

SIP Proxy

GW-B

PBX

INVITE

INVITE

Setup

Setup

100 Trying

Call Proceeding

100 Trying

100 Trying

Call Proceeding
Alerting

Call Proceeding
Alerting

180 Ringing

180 Ringing

180 Ringing

Alerting

Connect

Connect

200 OK

200 OK

200 OK

Connect
Connect Ack

ACK
Connect Ack

ACK
Voice
Disconnect

RTP

Connect Ack
Voice

RTP
BYE

Voice
Disconnect
Release

BYE
Release

Disconnect
Release

200 OK
Release Complete

200 OK
Release Complete

Release Complete

Callmanager acting as B2BUA

SIP Phone
INVITE, with SDP
100 Trying

CCM

GW-B

INVITE
183 Session Progress, with SDP
Session Progress, with SDP
200 OK, with SDP
ACK, with SDP
200 OK, with SDP
ACK
RTP
BYE
200 OK
BYE
200 OK