sip and voip state of the nation internet2 sip.edu workshop minneapolis, minnesota feb 2007
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SIP and VoIP State of the Nation Internet2 Sip.EDU Workshop Minneapolis, 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. - PowerPoint PPT PresentationTRANSCRIPT
SIP and VoIPState of the Nation
Internet2 Sip.EDU WorkshopMinneapolis, Minnesota
Feb 2007
Walt Magnussen, Ph.DTexas A&M UniversityDirector TAMU ITEC
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
cent
rex-
styl
e fe
atur
es
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
usuallyused 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
01020304050607080
1999 2000 2001 2002 2003 2004 2005 2006
SIPSIPPINGSIMPLE
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
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)
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– [email protected]– 979-845-5588