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SIPPING - IEPREP Joint Meeting. Fred Baker - IEPREP co-chair Rohan Mahy - SIPPING co-chair. SIP Architecture review. Proxies are call stateless -- in the Internet proxies don’t have any linkage with QoS, or call counting
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SIPPING - IEPREP Joint Meeting Fred Baker - IEPREP co-chair Rohan Mahy - SIPPING co-chair
SIP Architecture review • Proxies are call stateless -- in the Internet proxies don’t have any linkage with QoS, or call counting • Proxies are fast -- standardizing prioritization inside a proxy has questionable value (implementation specific) • User Agents (Gateways, Phones, Media Servers) are responsible for requesting priority or preemption and “doing the right thing” (ex: preempting focus, requesting QoS) when receiving it
Making a call in a hybrid network PSTN “Internet” Issues: Policy in call acceptance: call waiting, preemption (SIP) Policy in bandwidth admission preferential reservation of bandwidth (RSVP) Exchange with PSTN: encoding and values PSTN
SIP-to-PSTN security requirements for prioritized access to PSTN resources Henning Schulzrinne Columbia University
Security requirements • End-to-end strong authentication and authorization of caller • not just theft of service, but system stability/performance issue • Intermediate (proxy?) authentication • delegate responsibility • not all VoIP gateways may be authentication-capable (many aren't) • Need to authorize any assertion of priority and authenticate the originator
Security requirements • Cross-domain • IP endpoint may be in different admin. domain than gateway • Require secrets not to be pre-installed • useability from any device • Authentication of PSTN gateway • desirable; required?
Privacy requirements • Call content • very likely separate docs • Signaling (resource and/or call setup) • reveals communication relationships • cannot rely on hop-by-hop • Fact of IEPREP call • sensitivity likely same (or lower) as call signaling
Other open requirements questions • Call routing based on support of resource priority • support of specific IEPREP namespace? • SIP URI of gateway enough? • or "call # using FOO priority service" no need to know gateway address • avoid two-stage dialing • Caller needs to discover support for namespaces • may require different authentication
System assumptions • What do we assume about the IP side? • purpose-built: require certain capabilities (signaling, resource reservation, security, ...) • any network: use SIP application on standard platform or plug in own SIP phone • no network changes • firewalls may not allow protocols beyond SIP and RTP • any SIP (pay) phone • no modifications to SIP phone • not much beyond two-stage dialing possible?