SIP and VoIPState of the Nation

Internet2 Sip.EDU WorkshopMinneapolis, Minnesota

Feb 2007

Walt Magnussen, Ph.D

Texas A&M University

Director TAMU ITEC

Agenda

• Standards

• Platforms

• Services

Standards

• The following slides are provided by Dr. Henning Schulzrinne, Columbia University

Evolution of VoIP

“amazing – thephone rings”

“does it docall transfer?”

“how can I make itstop ringing?”

1996-2000 2000-2003 2004-

catching upwith the digital PBX

long-distance calling,ca. 1930 going beyond

the black phone

SIP is PBX/Centrex readycall waiting/multiple calls

RFC 3261

hold RFC 3264

transfer RFC 3515/Replaces

conference RFC 3261/callee caps

message waiting message summary package

call forward RFC 3261

call park RFC 3515/Replaces

call pickup Replaces

do not disturb RFC 3261

call coverage RFC 3261

from Rohan Mahy’s VON Fall 2003 talk

simultaneous ringing

RFC 3261

basic shared lines dialog/reg. package

barge-in Join

“Take” Replaces

Shared-line “privacy”

dialog package

divert to admin RFC 3261

intercom URI convention

auto attendant RFC 3261/2833

attendant console dialog package

night service RFC 3261

centr

ex-s

tyle

featu

res

boss/admin features

attendant features

IETF VoIP efforts

SIP(protocol)

SIPPING(usage, requirements)

ECRIT(emergency calling)

AVT(RTP, SRTP, media)

ENUM(E.164 translation)

IPTEL(tel URL)

SIMPLE(presence)

GEOPRIV(geo + privacy)

usesmay use

uses

provides

usually

used with

IETF RAI area

MMUSIC(SDP, RTSP, ICE)

XCON(conf. control)

SPEERMINT(peering)

uses

SPEECHSC(speech services)

SIGTRAN(signaling transport)

uses

A constellation of SIP RFCs

Resource mgt. (3312)Reliable prov. (3262)INFO (2976)UPDATE (3311)Reason (3326)SIP (3261)

DNS for SIP (3263)Events (3265)REFER (3515)

DHCP (3361)DHCPv6 (3319)

Digest AKA (3310)Privacy (3323)P-Asserted (3325)Agreement (3329)Media auth. (3313)AES (3853)

Non-adjacent (3327)Symmetric resp. (3581)Service route (3608)User agent caps (3840)Caller prefs (3841)

ISUP (3204)sipfrag (3240)

Security & privacy

Configuration

Core

Mostly PSTN

Content types

Request routing

SIP, SIPPING & SIMPLE –00 drafts

includes draft-ietf-*-00 and draft-personal-*-00

0

10

20

30

40

50

60

70

80

1999 2000 2001 2002 2003 2004 2005 2006

SIP

SIPPING

SIMPLE

RFC publication

0

2

4

6

8

10

12

14

2001 2002 2003 2004 2005 2006

SIP

SIPPING

SIMPLE

IETF WG: SIP

• ~ 44 SIP-related RFCs published

• Activities:– hitchhiker’s guide– infrastructure:

• GRUUs (random identifiers)

• URI lists• XCAP configuration• SIP MIB

– services:• rejecting anonymous

requests• consent framework• location conveyance• session policy

– security:

• end-to-middle security

• certificates

• SAML

• sips clarification

– NAT:

• connection re-use

• SIP outbound

see http://tools.ietf.org/wg/sip’/

IETF WG: SIPPING

• 31 RFCs published• Policy

– media policy– SBC functions

• Services– service examples– call transfer– configuration framework– spam and spit– text-over-IP– transcoding

• Testing and operations– IPv6 transition– race condition examples– IPv6 torture tests– SIP offer-answer examples– overload requirements

Conclusion• Core standards for media and signaling are finished

– can build PBX-equivalent devices and services on a large scale• see BT, FiOS, Vonage

• Lots of decent server implementations (various vendors; SER, openSER, Asterisk)– but lack of good soft clients for major OS platforms

• Ossification of Internet requires application complexity– kludge around NATs, lack of QoS– lack of credential infrastructure

• Intersection with policy and business models– NGN, 3G: maintain voice as high-value monopoly service

• Not a protocol engineering effort, systems engineering

Platforms

• Enterprise VoIP

• Carrier Platforms

• Open Source

Enterprise Platforms

• Cisco Call Manager– Still supporting Skinny with robust SIP support

• Trunk side (4.0 and greater)• Line Side (5.0 and greater)

• Nortel CS-1000 BCM and MCS 5100– Unistim for full feature functionality– Trunk side SIP only at this time (4.0 and greater)

• Avaya– Sip support for line and trunk

• 3COM– Built on SIP

Carrier Platforms

• Exclusively SIP

• Major vendor support– Broadsoft (Verizon and many TISP providers)– Cedar Point– Sonus

Open Source

• Being implemented in many startup solutions (i.e. DetD) – SIP only as well– ASTERISK– IPTel SER– Open SER– Sip Foundry SIPX

• Campus wide solutions– Penn State– UNC– Sam Houston State (Texas)

Services

• Trunking

• Peering

• Hosted Centrex

• E-911

• Peer to Peer

Trunking

• IP trunk access to PSTN avialable from several service providers today (Level3, Paetec etc.)– Inbound and outbound LD– Local access under LNP– 800 Services– Directory Assistance

• Advantages– Allows converged access– Lower cost– Eliminates Local PRI costs– Allows easy diversity (when coupled with LCR)

Peering

• Commonly used by VoIP service providers– Uses ENUM for scalability– Various service providers

• Voice Peering Forum http://www.thevpf.com/ • Verisign NRD service

– Hardware solutions can augment service• Nextone does MOS calculations and dynamic

rerouting when necessary.

Hosted Centrex

• Talked about a lot by Industry

• Currently offered by Verizon HIPC

• Under evaluation at TAMU (more to follow)

E-911

• Location of devices not required on most campuses (but highly recommended)

• Fixed locations supported by:– Telemanagement systems (i.e. Pinnacle)– ILEC services (Verizon and AT&T)– Hardware solutions (i.e. Cisco)

• TAMU solution– Lock down fixed telephones per port– ID mobile devices (softphones) in database

• More elequent solutions to appear under NG911

Peer to Peer

• Not supported by most campuses but rampant (i.e. Skype)– Blocked by many international service

providers– UC Santa Barbra approach is to block

http://www.oit.ucsb.edu/connect/skype.asp

Questions ?

• Contact info:– Walt Magnussen, Ph.D.– ITEC Director– telecom@tamu.edu– 979-845-5588