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Core developer for Jiyo (https://www.jiyo.com) for b2b unified communication , media streaming and API platform development . Part of innovation and R&D telecom ...
Missing: q= 3A% 2Ftelecom. 2Fcategory% 2Fmedia-
Encapsulting Protocols · Multihoming protocols and mobility · HTTP/2 – offer/answer signaling for WebRTC call · Websockets as VOIP signal transport medium · PJSIP ...
Missing: q= 3A% 2Ftelecom. 2Fcategory% 2Fmedia-
This article lists some external programmable Call Control APIs, internal APIs for biling , health as well as Rate limitting. Public API endpoints. Programmatic ...
Missing: q= 3A% 2Ftelecom. 2Fcategory% 2Fmedia-
Call processing framework ( CPF). Signalling and call routing logic, mostly in SIP and trunks. Manages identities such as Call Line information, Called Party ...
Missing: 2Ftelecom. 2Fcategory% 2Fmedia-
Specialized in CPaaS, carrier-grade WebRTC-SIP telecom platforms for Unified communication-collaboration, signaling gateways, SBC, soft turrets, ...
Missing: 3A% 2Ftelecom. 2Fcategory% 2Fmedia-
Session Border Controllers ( SBC ) assist in controlling the signalling and usually also the media streams involved in calls and sessions. They are often part ...
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With this article I will outlines the SIP servlet creation and various call routing logic development. A simple proxy SIP ser vlet application also has 4 parts ...
Missing: q= 3A% 2Ftelecom. 2Fcategory% 2Fmedia-
Feb 2, 2019 · Why is NAT is important in SIP? Types of NAT solutions; NAT behaviours; RTP NAT; Fixing NAT; NAT Traversal Module. Why use keepalive when ...
Missing: q= 3A% 2Ftelecom. 2Fcategory% 2Fmedia-
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