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Published byNathaniel Page Modified over 9 years ago
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17 February 2016 SIPPING - IEPREP Joint Meeting Fred Baker - IEPREP co-chair Rohan Mahy - SIPPING co-chair
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17 February 2016 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
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17 February 2016 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
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17 February 2016 SIP-to-PSTN security requirements for prioritized access to PSTN resources Henning Schulzrinne Columbia University
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17 February 2016 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
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17 February 2016 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?
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17 February 2016 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
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17 February 2016 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
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17 February 2016 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?
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