SIP, SDP and VoIP

David A. Bryan

CSCI 434/534

December 6, 2003

VoIP

• VoIP – Voice over IP (Internet Protocol)• Basic idea is to run telephone type

services (and sometimes video) over IP– Cost savings (administration, arbitrage)– Tax/tariff avoidance– Security (internal, encryption)– Advanced features

• Next generation (3G and later) cellular phones run over IP networks

Telephony Protocols

• Original phone network is circuit switched– SS7 is the protocol used for this – NOT a

computer standard– Defined by ITU – International

Telecommunication Union

• VoIP is packet switched – uses computer networks– H.323 is also an ITU standard, but designed

to work over packet switched networks

Telephony Protocols

• SIP– IETF defined a protocol for VoIP, called SIP– Designed for packet networks, useful for any

sort of multimedia session establishment– Support for integration with email/web

services– SIMPLE is a set of extensions to SIP to

support instant messaging

H.323 vs. SIP

Feature H.323 SIP

Encoding Binary Text

Control Centralized server Endpoints (using proxy)

Intelligence Most in central server; core

Most in the endpoints; edge

Modularity Monolithic (all in one server)

Can be multiple servers

Media RTP RTP

Media (RTP)

SIP Call

SIP ServerSIP/SDP SIP/SDP

SIP

• Session Initiation Protocol• Defined (mostly) in IETF RFC 3261

– Many other drafts extend SIP and add new features

• Text based, very similar to HTTP• Sets up a multimedia session between endpoints• Allows mobility – locate users using a URL

(sip:user@sip.cs.wm.edu)• Usually runs on port 5060• Can use either UDP or TCP as transport• Allows for IPv4 and IPv6. Next generation cellular

phones (Europe) use SIP over TCP/IPv6

SDP

• Session Description Protocol• SIP doesn’t try to describe the media stream• Defined in IETF RFC 2327• This is used by SIP to actually describe the type

of media being used, where to send it, etc. – Included in SIP message as a payload

• This is an older IETF protocol– IETF tries to promote reuse whenever possible

• Yet another protocol, RTP, is used for the actual media

SIP

• Logically, there are several parts to a SIP system– UA (User-Agent, Endpoint) – the phone itself – can

also be a softphone – computer software– Registrar – keeps track of where the user is within a

system– Redirect Server – used to inform devices when they

need to contact different locations– Proxy Server – used to relay messages back and

forth within the system• In practice, several functions may actually be in

the same server/program

Registration

“I’m Alice, and I’ll be using aphone at 192.168.0.1 today!”

192.168.0.1 SIP Server

sip.phonecompany.com

When I get a call for Alice, I’ll know to

contact her at 192.168.0.1!

10.0.1.1

“Lunch time! Use my mobileat 10.0.1.1”

Ah, now I will have to use 10.0.1.1 to reach Alice.

Proxy

sip.phonecompany.com

sip.quickmart.com

“I need to call Bob!”

“Call here for Bob from Alice”

“Call for youfrom Alice!”

Bob is over at quickmart.com. I’ll proxy the

call over there!

Redirection

sip.phonecompany.com

sip.quickmart.com

“I need to call Bob!”

“Call here for Bob from Alice”

“Call for youfrom Alice!”

sip.diveapartment.com“Try him at diveapartment.com”

“Call here for Bob from Alice”

SIP Messages

• SIP messages fall into two types – requests and responses– Requests include a method telling what it is they are

doing. – REGISTER, to register a phone or device– INVITE, to start a new session (call)– BYE, to end a session that is established– CANCEL, to end a session that is still being set up

(no final response yet)– ACK, part of the triple handshake

SIP Messages

• Responses are numeric, much like HTTP– 1xx Provisional – 100 Trying, 180 Ringing– 2xx Successful – 200 OK– 3xx Redirection – 301 Moved Permanently,

302 Moved Temporarily– 4xx Failure – 404 Not Found, 410 Gone, 403

Forbidden– 5xx Server Failure – 503 Service Unavailable– 6xx Global Failure – 600 Busy Everywhere

RegistrationSIP Server

REGISTER

401 Unauthorized

REGISTER

200 OK

Basic CallSIP Server

INVITEINVITE

100 Trying

180 Ringing180 Ringing

200 OK200 OK

ACKACK

BYEBYE

Bi-directional Media between endpoints

200 OK200 OK

No answerSIP Server

INVITEINVITE

100 Trying

180 Ringing180 Ringing

487 Request Terminated487 Request Terminated

200 OK200 OK

CANCEL

200 OK

CANCEL

200 OK

INVITE messageINVITE sip:bob@quickmart.com SIP/2.0 Via: SIP/2.0/TCP aliceua.phonecompany.com:5060From: Alice <sip:alice@phonecompany.com> To: Bob <sip:bob@quickmart.com> Call-ID: 3848276298220188511@phonecompany.com CSeq: 1 INVITE Contact: <sip:alice@aliceua.phonecompany.com;transport=tcp> Content-Type: application/sdp Content-Length: 151

v=0 o=Cisco-UserAgent 10010 605 IN IP4 aliceua.phonecompany.com s=SIP Callc=IN IP4 192.0.1.1 t=0 0 m=audio 49172 RTP/AVP 0 8 16

INVITE messageINVITE sip:bob@quickmart.com SIP/2.0

SIP request, method is INVITE – this message will start a new callVia: SIP/2.0/TCP aliceua.phonecompany.com:5060

Via header tracks where it has been – so far just Alice’s UAFrom: Alice <sip:alice@phonecompany.com>

Message is from Alice, and this is the SIP URL for AliceTo: Bob <sip:bob@quickmart.com>

Alice is trying to contact Bob at his SIP URLCall-ID: 3848276298220188511@phonecompany.com

Call-ID is a (hopefully) unique ID to track this particular call. Alice’s UA generated this

CSeq: 1 INVITE We also count messages within a call to see what responses gowith what messages. This first exchange is an INVITE exchange.The response to this will also have a CSeq of 1 INVITE

Contact: <sip:alice@aliceua.phonecompany.com;transport=tcp> How to contact Alice directly – you can sometimes “skip” the proxy.

<snip!>

INVITE message<snip!> Content-Type: application/sdp Content-Length: 151

The content of the message is SDP – to describe the media of thecall. We have 151 bytes of SDP

v=0 v is the SDP version number -- 0

o=Cisco-UserAgent 10010 605 IN IP4 aliceua.phonecompany.com o is a description of the UA and a session ID

s=SIP Calls is an string to encode a name for this session

c=IN IP4 192.0.1.1 c tells the other side what address to send the media to

t=0 0 t indicates start and end times. 0 and 0 don’t specify times in advance

m=audio 49172 RTP/AVP 0 8 16m describes the media. We are listening for audio of type RTP/AVP on port 49172

Network Architecture

• Several ways to have a device that connects to an IP network– True IP endpoint – a phone w/an ethernet connection

– often quite expensive– ATA – Analog Telephone Adaptor (this is what you

get if you sign up with Vonage) – plugs into the ethernet, and you plug a phone into it

– IP enabled PBX (make an existing office phone system speak SIP)

– Soft Client – program that runs on a computer and is used with microphone and speakers

Network Architecture

• How about connecting to the outside world?– If you want to talk to another SIP user, your

proxy may know how to contact that proxy directly

– To call out (or to have calls come in) you use a gateway. It connects to one or more regular phone lines to let you receive and place calls to old fashioned “Plain Old Telephone Service”, or POTS

Gateway example

SIP Server

Gateway

SIP/IP network POTS network

Problems with VoIP

• Cost of entrance– while the maintenance and operating costs are low,

equipment can be expensive

• Reliability– Still issues with reliability of systems

• Safety/Security– 911 is a large issue, particularly routing to the right

place– Interception is difficult, and encryption is easy with

softphones – possibly a national security issue

Problems with VoIP

• NAT and firewall traversal– Media and signaling on different ports leads to

issues with media not being able traverse firewalls

– Imbedded addresses within SIP messages are a problem for NATs

• Taxes/regulation– Remove some incentive for VoIP

Open Source SIP projects

• Several good open source SIP projects– VOCAL

• Full SIP “softswitch” – includes proxy, softclient, voice mail, etc.

– ReSIProcate• Newer, much nicer stack than VOCAL, but few

applications so far

– IPTel Sip Express Router• High performance Proxy. Less full featured than

VOCAL, but faster and more robust

References

[1] SIP versus H.323, iptel.org, (http://www.iptel.org/info/trends/sip.html)

[2] SIP Tutorial, iptel.org, (http://www.iptel.org/sip/siptutorial.pdf)

[3] SIP RFC, RFC-3261, IETF, (http://www.ietf.org/rfc/rfc3261.txt)

[4] SDP RFC, RFC-2327, IETF, (http://www.ietf.org/rfc/rfc2327.txt)

[5] VOCAL Project, (http://www.vovida.org)[6] ReSIProcate Project, (http://www.resiprocate.org)[7] Vonage, (http://www.vonage.com)