SIP tells the end point that its phone should be “ringing ” SIP is used to agree on session attributes used by the calling and called party. SIP is used by end points to negotiate media capabilities, such as agreeing on a mutually supported voice codec. SIP is used by end points to determine whether they will “answer” a call. The table below shows the five major functions within SIP from a VoIP point of view.Įnd points (telephones) notify SIP proxies of their location SIP determines which end points will participate in a call. SIP also has great breadth as it does more than just handle call setup. In this document, we'll focus on SIP's capabilities for VoIP, and how it sets up calls that then use RTP (the Real-time Transport Protocol) to actually send the voice data between phones. For example, in addition to simple telephone calls, SIP can also be used to set up video and audio multicast meetings, or instant messaging conferences. It was designed to be a general-purpose way to set up real-time multimedia sessions between groups of participants. SIP is a very flexible protocol that has great depth. You may have also heard of H.323, an ITU protocol with similar function. In the world of VoIP, SIP is a call setup protocol that operates at the application layer. Once the call has been setup, the phones enter a "data transfer" phase of the call using an entirely different family of protocols to actually move the voice packets between the two phones. The first phase is "call setup," and includes all of the details needed to get two telephones talking. In IP and traditional telephony, network engineers have always made a clear distinction between two different phases of a voice call. We can't wait to help you build.SIP is the Session Initiation Protocol. Read about our SIP Trunking offerings, read about our SIP integration in our documentation, or talk to sales. Whether you're taking the first steps towards modernizing a call center, adding additional channels of support such as WebRTC, Chat, and SMS, or looking to reduce latency and leverage our massive scale, Twilio has a solution for you. How SIP and Twilio Can Work with Your Business We've also written more about SIP Trunking if you're itching to get started or just want to learn more. We offer unlimited concurrent call capacity - let us worry about the scaling and the capacity while you run your business. Connect to a SIP URI that's local to your firm and we will do the rest - all over a global private backbone that minimizes traversal on the public internet. Twilio also offers our Elastic SIP Trunking service, where we can instantly provision a SIP network for you worldwide. Twilio Interconnect also offers additional QoS guarantees and SLAs as well as enterprise-grade security. We have provisions to connect with your cloud infrastructure directly (such as with Amazon Web Services, Google's Compute Cloud, and Microsoft Azure), physically interconnect at one of our global locations, tunnel over a VPN, or connect to your MPLS Network. Twilio also offers Twilio Interconnect, a varied service where we offer a direct connection that can largely or wholly avoid the public internet. Here's our overview of connecting your existing SIP communications infrastructure to us. It's easy to connect your on-site or virtual Private Branch Exchange (PBX) with Twilio's servers. Using SIP to connect to Twilio is one way to get access to Twilio's global reach and powerful automation and scripting functions. * See Twilio's Best Practices for SIP Security * How Twilio Uses SIP SIP is a text-based protocol and uses UTF-8 encoding. Unlike HTTP, SIP can use UDP or other transport types. It was first standardized in March 1999's RFC 2543 and has been subsequently revised in June 2002's RFC 3261 (with various updates since then).īy mapping SIP endpoints ( user agents, in SIP terminology) to Uniform Resource Identifiers (URIs), SIP is quite similar to the HTTP model of requests and responses. An Easy to Implement Standard for Digital CommunicationsĬompared to predecessors such as H.323, SIP is a much easier to implement and much more flexible protocol. Most commonly, SIP is used for Voice Over IP (VoIP) services, but is also often used for other communications sessions such as video calls and instant messaging sessions. Session Initiation Protocol (SIP) is a signalling protocol for initiating, terminating, and modifying user sessions over an IP network.
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