SIP - The Key to VoIP

SIP – The Key to VoIP

1/25copyright © 2015 by schlager communications services GmbH

SIP - The Key to VoIP

ronaldschlager.com © Ronald Schlager



Overview

General

SIP Addresses

Message Elements

Transactions

Methods

Dialog

Use of DNS and Location Server



General

Design goals:

Distributed intelligence in different systems (end devices, servers, gateways,...)

Advantages:

CPU and memory consumtion in server are low

Reliability and scalability are high(no central device that may fail)

SIP protocol data units (PDU) use different routes than media streams



General

Runs over TLS (over TCP), TCP, UDP or SCTP: Selection of transport protocol is based on:

Indication in URI Selection by content in DNS record (NAPTR)

Following selection sequence has to be considered:

1. TLS over TCP („SIPS+D2T“)

2. TCP („SIP+D2T“)

3. UDP („SIP+D2U“)

4. SCTP („SIP+D2S“)Comment: D2x (x = T, U, S) names parts of DNS record entries



General

SIP solutions mostly use additional protocols:

Session Description Protocol (SDP)

To describe a session (by parameters)

See later

Stream Control Transmission Protocol (SCTP)

Reliable transport protocol similar to TCP

Originally developed for PSTN signalling over IP



General

Additional Protocols (cont.):

Real-time Transport Protocol (RTP) and Real-time Transport Control Protocol (RTCP)

To transport and to control media streams

Secure RTP and Secure RTCP

Encrypted transport of media streams

Domain Name Service (DNS), Dynamic Delegation Discovery System (DDDS)

For name resolution by means of links to lists of IP addresses



General

Additional Protocols (cont.):

Remote Authentication Dial In User Service (RADIUS)

For centralized authentification of users and devices

Lightweight Directory Access Protocol (LDAP)

Is directory service for centralized management of network and user objects

The coordination of all protocols allows the smooth operation of a SIP solution



SIP Addresses (URI)

Includes host address (is mandatory), optional user name, port number, different parameters

Can be used like web URL, with name resolution by Domain Name System (DNS)

Addressing concept very similar to

ITU-T E.164 calling numbers (tel:+432742319150)

Email addresses (mailto:...) or

HTTP URLs (http://...)

UA server registers address at Registrar (SIP REGISTER)

UA client uses address for session setup



SIP Addresses (URI)

Are globally defined (for global reachability)

General format scheme acc. RFC 3261:

sip:user:password@host:port;uri-parameters?headers

user:password: specific identification (user ID and password) of a user of a host

host: IPv4 or IPv6 address or (Fully Qualified) Domain Name

port: Port number where the request is to be sent

uri-parameters?headers: parameters affecting

a request constructed from the URI



SIP Addresses (URI)

Format Scheme acc. RFC 3986, Examples:

sip:[email protected] sip:[email protected]?

subject=call May also provide geo location information,

specifically interesting for e.g. emergency calls

<gml:coordinates>33.001111N 96.68142W</gml:coordinates>

Format scheme for secure signalling (lt. RFC 3261):

sips:[email protected] Requires Transport Layer Security (TLS) over TCP

for encrypted transport of SIP



SIP Addresses (URI)

2 Types of URI:

„Address of Record“ (AOR):

sip:[email protected] (requires SRV record in DNS server for address resolution to find SIP server in domain ronaldschlager.com)

„Fully Qualified Domain Name“ (FQDN) or „Contact“ (identifies a device within a domain)

sip:[email protected] or sip:[email protected] (unique server name)

No indication of a distinct server

Exact specification ofSIP server SIP1



Message Elements

Example for Message: INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP SIP1.provider.at:5060;

branch=u4eF3o5g3Z9; Max-Forwards: 70 Contact: <sip:[email protected]> To: „Ronald“ <sip:[email protected]> From: „Peter“

<sip:[email protected];tag=3746528393 Call-ID: [email protected] Cseq: 1 INVITE Date: Wed, 30 April 2003 11:17:56 GMT Content-Type: application/sdp Content-Length: 132



Message Elements

Following information will be exchanged for each session:

Request line: “INVITE”: SIP method name or SIP request type (here: session setup request)

Request URI: sip:[email protected] Indicates the destination (called party)

“SIP/2.0”: SIP Version 2

“Via”: source or caller address (also: proxy address),optional port number where to send the responses

Additional “Via” lines may be inserted by each Proxy who wants to forward responses



Message Elements

Following information will be exchanged for each session (cont.):

“Max. Forwards”: limits the max. number of “hops” (Proxies), will be decremented (minus 1) by each forwarding Proxy (like “Time To Live” field in IP)

“Contact”: SIP or SIPS URI (as fully qualified domain name, FQDN)

Represents a direct route to a contact, use of an IP address allowed

Indicates where to send further requests

Exception: if Proxy indicates request to forward all messages (with “record route”)



Message Elements


“To”: display name (acc. RFC 2822) and SIP or SIPS URI of the original destination

“From”: display name and SIP or SIPS URI of the originator of the request

Tag parameter: is random string, used for identification purposes of end devices

“Call-ID”: globally unique identifier for this call, generated be combination of random number and IP address or hostname



Message Elements


“CSeq” (Command Sequence): integer and method name, is incremented for each new request and is a traditional sequence number, should prevent “replay”-attacks

“Content Type”: description of the message body (text, plain, application,...)

“Content Length”: octet count of the message body



Message Elements

Message body is separated from SIP header by CRLF

Message body may contain:

Description of media capabilities(Session Description Protocol)

Instant message (compare with ISDN User-to-User Signalling)

JPEG

MIME



Transactions

Basic principle:

A B

Provisional Response

RequestCreate Transaction Create

Transaction

Final ResponseDestroy

TransactionDestroy

Transaction



Transactions

Defined requests:

REGISTER Registrion at registrar

INVITE Initiates handshake

ACK Acknowledges sucessful handshake

BYE Terminates session

CANCEL Cancels previous, not closed request

Queries capabilities of peersOPTIONS



Transactions

Defined Message Codes (Response Classes): Codes „based“ on HTTP („xyz explanation“) Receiver must understand „x“

Provisional Status Code (100-199)

Final

Successful (200-299)

Redirection (300-399)

Request Failure (400-499)

Server Failure (500-599)

Global Failures (600-699)



Methods

Invite: invitation for session establishment with a UA

Invite re-invite: modifying an existing session

Register: registering a location to the registrar server

Ack: to confirm a request

Cancel: to cancel an invite

Bye: to terminate a session

Options: to request supported functions and features of partners without the need for session setup



Method Extensions (Examples)

Info: used for exchange of control information for a session (e.g. in ISDN signalling to control a set of telephony services)

Update: allows a client to modify session parameters, e.g. new codecs, stream formats,...)

Message: used for instant messaging beween clients

Refer: originator of a refer will be informed about availability of requested resources (devices, functions,...)

Prack: provisional acknowledgement, similar to ACK, passes proxys and is controlled by them

Message Waiting Indication



A B

Dialog

A BCreateDialog Create

DialogCreate

DestroyDialogDestroy

DialogDestroy

Update Update/ModifyDialog

Update/ModifyDialog

Update Update/ModifyDialog

Update/ModifyDialog



Use of DNS and Location Server

Location ServerDNS Server

Inboand Proxy Server Outboand Proxy Server

IP phoneUser A (Caller)

IP phoneUser B (Called)

SIP SIPSIP

Media stream

DNS ?



Ronald Schlager´s Resources

Many thanks for your visit.

Visit Ronald Schlager´s resources for more information on ITU-T H.323 and SIP protocol details:

This presentation is part of courseware: „SIP – The Key to VoIP“

Web:

Ronald Schlager´s Blog on Business VoIPwww.ronaldschlager.com

http://www.ronaldschlager.com/eBooks4.htm





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SIP - The Key to VoIP