@techreport{rosenbergjennings-dispatch-ript-00,
    number =    {draft-rosenbergjennings-dispatch-ript-00},
    type =      {Internet-Draft},
    institution =   {Internet Engineering Task Force},
    publisher = {Internet Engineering Task Force},
    note =      {Work in Progress},
    url =       {https://datatracker.ietf.org/doc/draft-rosenbergjennings-dispatch-ript/00/},
    author =    {Jonathan Rosenberg and Cullen Fluffy Jennings and Anthony Minessale and Jason Livingood and Justin Uberti},
    title =     {{RealTime Internet Peering for Telephony}},
    pagetotal = 42,
    year =      2020,
    month =     feb,
    day =       7,
    abstract =  {This document specifies the Realtime Internet Peering for Telephony (RIPT) protocol. RIPT is used to provide peering of voice and video communications between entities. These include a traditional voice trunking provider (such as a telco), and a trunking consumer (such as an enterprise PBX or contact center), or between a video conferencing endpoint deployed in an enterprise, and a video conferencing SaaS service. RIPT is an alternative to SIP, SDP and RTP for these use cases, and is designed as a web application using HTTP/3. Using HTTP/3 allows implementors to build their applications on top of cloud platforms, such as AWS, Azure and Google Cloud, all of which are heavily focused on HTTP based services. RIPT also addresses many of the challenges of traditional SIP-based peering. It supports modern techniques for load balancing, autoscaling, call-preserving failover, graceful call migrations, security by default, STIR-based caller ID, provisioning, and capabilities - all of which have been challenges with traditional SIP peering and voice trunking. Since it runs over HTTP/3, it works through NATs and firewalls with the same ease as HTTP does.},
}