AARNet Copyright 2011
SIP Deep Dive
Network Operations
Bill Efthimiou
APAN33 SIP workshop
February 2012
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Agenda
1.
2.
3.
4.
5.
6.
7.
8.
9.
10.
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SIP Basics
SIP Components and SIP Addressing
SIP Messaging Syntax
SIP Transaction and Dialog
SIP Call Flows
DNS NAPTR and SRV
SIP Header
SIP Mobility
SIP and B2BUA – Back to Back User Agent
SIP Peering
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The Basics of SIP
• RFC3261
• Session Initiation Protocol
• Application-layer Signalling protocol
• Setup, Modify and Tear down sessions
• Invite participants to existing sessions
• Add/remove media to/from existing sessions
• Establish user Presence
• Personal mobility – single identity
• IPv4 and IPv6 support
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Where does SIP fit in?
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SIP User Agents
• User Agent Client (UAC) – An entity initiates a call
• User Agent Server (UAS) – An entity receives a call
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SIP Servers
• Registrar Servers – users/services registration
• Location Servers – maintaining users locations
• Redirect Servers
 Accept request, map the requested address to new addresses, and return to
the client
 No initial requests
 Do not accept calls
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•Proxy Servers – next slide
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SIP Proxy Server
•Proxy Servers – forwarding requests, rewrite message, forking, etc
 Stateful – remembers all requests
 Stateless – forgets requests after forwarded
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SIP Addressing
SIP Addresses are URIs with the same format as an email
address i.e. user@domain or user@host. They can refer to
Services, Devices and Users.
• Service – URI points to conference,
sip:61262112650@aarnet.edu.au is a conference service on the
MCU (bridge)
• Devices – URI represents a single device/application, sip:sydfax1@aarnet.edu.au is a fax machine
• Users – URI represents a user’s contact detail,
sip:leon.li@aarnet.edu.au
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SIP Registration
• A Common operation in SIP. Registration is a common way for a server
to learn the current location of a user.
• REGISTER messages associate a users SIP or SIPS URI (eg,
sip:bob@biloxi.com) with the machine/phone/etc into which he/she is
currently logged in.
• The registrar writes this association, also called a binding, to a database,
called the location service, where it can be used by the proxy in the
domain. Often, a registrar server for a domain is co-located with the
proxy for that domain.
• A user “Bob” is not limited to registering from a single device. For
example, both his SIP phone at home and the one in the office could send
registrations. This information is stored together in the location service
and allows a proxy to perform various types of searches to locate Bob.
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SIP Messages
There are two types of SIP Messages
 Requests
 Responses
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SIP Messages – Request-Line, example
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SIP Messages – Status-Line, example
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SIP Requests
• INVITE – to initiate a session
• ACK – to indicate the receipt of the corresponding msg
• BYE – to terminate a session
• CANCEL – to cease a previous request
• REGISTER – to send registration details to a Registrar
• OPTIONS – to query another UA or a proxy server as to its capabilities
• INFO – to carry session related control information
• REFER – to direct the recipient to contact a 3rd party, e.g. call transfer
• PRACK – provisional ACK (to 1XX only except 100 Trying)
• PUBLISH – to advertise event state, like presence
• SUBSCRIBE – to request the status of a session/UA etc
• for a detailed list,
http://en.wikipedia.org/wiki/List_of_SIP_request_methods
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SIP Responses
• Response Code (detailed list at
http://en.wikipedia.org/wiki/List_of_SIP_response_codes)
Description
Code Example
Informational
1XX
100 Trying
180 Ringing
Success
2XX
200 OK
Redirect
3XX
300 Multi Choices
301 Move permanently
Clients Errors 4XX
400 Bad Request
401 Unauthorized
404 Not Found
408 Timeout
483 Too Many Hops
Server Errors
5XX
500 Internal Server Error
503 Service Unavailable
Global Errors
6XX
600 Busy Everywhere
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301 Moved Permanently
The user can no longer be found at the address in the Request-URI,
and the requesting client SHOULD retry at the new address given by
the Contact header field. The requestor SHOULD update any local
directories, address books, and user location caches with this new
value and redirect future requests to the address(es) listed.
Source: RFC3261
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302 Moved Temporarily
The requesting client SHOULD retry the request at the new address(es)
given by the Contact header field. The Request-URI of the new request
uses the value of the Contact header field in the response.
The duration of the validity of the Contact URI can be indicated through
an Expires header field or an expires parameter in the Contact header
field. Both proxies and UAs MAY cache this URI for the duration of the
expiration time. If there is no explicit expiration time, the address is only
valid once for recursing, and MUST NOT be cached for future
transactions.
Source: RFC3261
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SIP Transactions and Dialogs
• Transaction: from the first
request to the final response
• Dialog: starts with an
INVITE transaction and ends
with a BYE transaction
• DialogID = Call-ID + local
tag + remote tag.
Note, From/To/Call-ID never
changed in a dialog.
Source: Building Telephony Systems with OpenSER, by Flavio E. Goncalves
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SIP Call Flow
Source: sipschool.com
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SIP call Routing- Fixed (How does a proxy server know where to forward the INVITE?)
Hard coding/Fixed Dial plan
example in the lab,
Dialled Number
Destination
1XXX
192.168.1.21
2XXX
192.168.1.22
3XXX
192.168.1.23
4XXX
192.168.1.24
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SIP call Routing- SRV RR
SRV Record, RFC 2782
– A specification of data in the DNS defining the location
_service._proto.name
TTL
class
SRV
priority
weight
port
target
_sip._udp.example.com
86400
IN
SRV
0
5
5060
1.1.1.1
_sip._tcp.example.com. 86400 IN SRV 10 60 5060 bigbox.example.com.
_sip._tcp.example.com. 86400 IN SRV 10 20 5060 smallbox1.example.com.
_sip._tcp.example.com. 86400 IN SRV 10 10 5060 smallbox2.example.com.
_sip._tcp.example.com. 86400 IN SRV 10 10 5066 smallbox2.example.com.
_sip._tcp.example.com. 86400 IN SRV 20 0 5060 backupbox.example.com.
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DNS NAPTR and SRV (Cont’d)
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SIP call Routing- NAPTR (using ENUM)
The example NAPTR records below show a translation from the number
range +123456* to sip:123456*@sample.edu in the SIP case.
For H.323 the NAPTR records map h323:123456* to
h323:123456*@sample.edu.
Source: NRENUM.net:
https://confluence.terena.org/display/NRENum/How+to+use+NAPTR+and+SRV+records+for+H.323+or+SIP+with+nrenum.net
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Example
DNS SERVER
Source: sipschool.com
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SIP Headers
SIP Header is HTTP like and contains important information.
Examples
– Registration
– Invite
• Next hop?
– Record-route
– Via
– Contact
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SIP Header Reminder
• Record-Route
Added by the proxy so that it can stay in the messaging path between the
endpoints for the duration of the session
• Branch
– To identify the transaction
– MUST always begin with the characters "z9hG4bK".
• Tag
Random string added by the UA for identification purpose
• Remember in the Reply header
–
–
–
–
Via
From
To
Call-ID
are copied exactly from Request.
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SIP Mobility
• Proxying and redirecting requests to a user’s current location
• Device/network independent
 PC, Smartphone, IP phone etc
 LAN, WiFi, 3G etc
• User must register
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SIP Call Forking - Parallel
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Source: sipschool.com
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SIP Call Forking - Sequential
Source: sipschool.com
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SIP B2BUA
• Back to Back User Agent (for example, SBC)
A logical network element operates between both end points of a phone
call or communications session and divides the communication channel
into two call legs and mediates all SIP signalling between both ends of
the call, from call establishment to termination. As all control messages
for each call flow through the B2BUA, a service provider may
implement value-added features available during the call.
• Benefit
 Centralise call management
 Billing
 Network topology hiding
 H.323 and SIP inter-working
 QoS enforcement
 NAT traversal
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SIP Trunking
• What is SIP trunking? A “logical” connection from a PBX from
a customer site to an Internet Telephony Service Provider
(ITSP) network.
• Benefit
• replace existing E1/T1
• flexible with number of channels
• number porting
• single existence for multiple regions (for example, SIP lines
for Sydney and Melbourne can co-located in Sydney)
• Rich Media can be enabled, such as rich callerID, video,
presence, etc.
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SIP Peering
• What is SIP peering? A “logical” connection between two
organizations
• Benefit
• lower cost
• trusted relationship
• AARNet’s use case
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Thanks
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