sip explained gary audin delphi, inc. [email protected] sponsored by
TRANSCRIPT
SIP ExplainedGary AudinDelphi, Inc.
www.telecomreseller.com
Speaker Background• Communications and security consultant
for 36 years• Speaker at Enterprise Connect, ITExpo and
100s of user conferences• Article and blog sites:
o www.nojitter.como www.webtorials.como www.telecomreseller.como www.networkworld.com o www.acuta.orgo www.searchunifiedcommunications.com
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SIP TrunkingPresentation
• “Successful SIP Trunking”• Wednesday 5:30 and Thursday 1:00
PM• Telecom Reseller theater exhibit #• Presenters: Gary Audin and Steve
Johnson
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Session Initiation Protocol
• A protocol is an information exchange procedure, a set of rules.
• SIP is a protocol to establish , manage, and terminate a connection (session) that is media independent.
• SIP is not specifically designed for digital voice.
• SIP operates over an IP network.
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SIP Does This• It locates the user and determines which end system
will be used in the proposed session.• It then learns the user's availability (is the user busy;
can he/she be disturbed?).• It determines the capabilities available at the user
end system for the session such as what media is supported.
• It establishes the session.• It manages the session, handling call termination, call
transfer, changes to session parameters, and so forth.• It is a peer-to-peer protocol running over UDP and TCP.
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SIP Features• User location can determine the end system to be
used for communication• User availability determines the willingness of the
called endpoint to engage in communications• User capabilities can determine the media and
media parameters to be used• Session setup endpoint ringing, establishment of
session parameters at both called and calling endpoints
• Session management including session transfer and termination, changing session parameters, and invoking services
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What’s in a SIP Session• “Session” = exchange of data between an
association of participants• Users can move among endpoints• Users may have multiple names and addresses• Users may communicate in different media• SIP enables internet endpoints:
o To discover each othero To characterize the session
• The location infrastructure supports name mapping and redirection services
• Endpoints can add/remove participants from session• Endpoints can add/remove media from session
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Not Part of SIP • SIP is not a vertically integrated
communications system. It is ONLY a component.
• SIP is independent of the services offered. o SIP provides mechanisms that can be
used to implement different services. o SIP can locate a user and deliver content
to the user’s current location.• SIP does not offer conference control
services nor prescribe how a conference is to be managed. 8
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SIP Does Not• SIP does not define the media carried
(voice, video, IM, data, games, graphics, photos…)
• SIPPING 19 defines a minimum set of telephony features not SIP
• SIP trunks are not able to provide interoperability between different vendors’ IP PBXs
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SIP Components • SIP is built upon a client/server architecture• User Agents (SIP Phones, SIP PCs, other
endpoints) • Servers (Used to locate SIP users or to
forward messages) • SIP Gateways:
To PSTN for telephony interworkingTo H.323 for IP Telephony interworking
• Client - originates message• Server - responds to or forwards message
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Multimedia Protocol Stack
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Application Layer
Physical/Link Layer
Internet Layer
Transport Layer
Signaling Media Utility
IP
TCP UDP
SIPH.323 RTP DNS DHCP
Media CodingSDP
AALx PPP
ATM V.90 Ethernet MPLS
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SIP Signaling Paths
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TrunkGateway
PSTN
Analog
T1/E1
PRI
Access / MediaGateway
IP LAN / WAN
FAX
Phone
Modem
Softphone
Server/DNS/DHCP/TFTP
IP Phone © Delphi, Inc. 2014
SIP Trunk Benefits• Flexible provisioning• Provider competition and enterprise
leverage• Eliminate VoIP gateways• Reduced conferencing costs• Low cost or free international calling• On-Net free calling
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SIP Trunk Connections
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PSTN
Legacy PBX
VoIP Gateway SBC
T1/E1/PRI Connectio
ns
SIP Service Provider
IP PBX
SBC
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SIP Trunk Providers• Can connect using vendor specific versions of SIP
trunk software• Standard SIP trunk connection is via SIPconnect
(not a standard but an agreed upon recommendation from the SIP Forum http://www.sipforum.org/sipconnect)
• SIPconnect version 1.1 is the latest one• May limit the media carried:
o Voice onlyo Video maybeo Secure connections maybeo Fax maybeo 911 and E911 information maybe
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Session Border Controllers
• A firewall rule set while also map layer 5 to layer 7 addresses
• Intrusion detection and prevention• Denial-of-service (DoS) attack prevention• VPN separation for shared resources• SIP-Transport Layer Security (TLS) handshaking
for authentication and encryption of SIP signaling • Secure Real Time Protocol (RTP) support• Support for IPsec tunnels• Transcoding, or conversion between different VoIP
codec technologies
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Where Are the Problems?
Delphi, Inc.
SIP Trunk IssuesFirewall Problems:
– Can block SIP packets– Cannot translate IP packet addresses
TCP may be used instead of UDPOne way audioDropped connectionsCall transfer failureRegistration failure
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State of SIP• Relatively easy to implement• Has gained considerable vendor and provider
acceptance• Allows flexible service creation• Extensible and scalable• Wide range of supporting products and services• Does not make PSTN interworking easy• Will not solve all IP Telephony issues such as
QoS
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Resources• Online education and certification www.thesipschool.com • “Ten SIP Trunk Equipment License Issues That Can Ruin Your Day
(or Month)” http://www.webtorials.com/content/2013/03/ten-sip-trunk-equipment-license-issues-that-can-ruin-your-day-or-month.html
• “Avoiding SIP Trunking Equipment Problems” http://www.webtorials.com/content/2013/02/avoiding-sip-trunking-equipment-problems.html
• “How to avoid SIP Trunk Implementation Problems” http://www.webtorials.com/content/2013/01/how-to-avoid-sip-trunk-implementation-problems-1.html
• “Easy SIP Trunking; No Yet”, http://www.telecomreseller.com/2011/09/19/easy-sip-trunking-not-yet/
• “Is IPv6 Good for VoIP? http://www.webtorials.com/content/2013/01/is-ipv6-good-for-voip.html
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