Voice over internet protocol VoIP Voice over internet protocol Amirhossein Saberi - 2017

Voip & protocols SIP, H.323, IAX MGCP, Megaco RTP, RTCP, SRTP, SDP Voice over Internet Protocol (Voice over IP, VoIP and IP telephony) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. The terms Internet telephony, broadband telephony, and broadband phone service specifically refer to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).

Voip vs traditional landline Caller ID, Call waiting, Video, Conference call, FAX, No limitation, No physical equipment LCR Cost

Voip equipments ATA PBX Servers IPPhone PSTN Interface cards Appliance VoIP Gateways Conference phones SBC

Media gateway & protocols

Integrated Services Digital Network (ISDN) PRI E1 (32c – 64 Kb) T1 (24c – 64 Kb) B-channels, D-channels Integrated Services Digital Network (ISDN) is a set of communication standards for simultaneous digital transmission of voice, video, data, and other network services over the traditional circuits of the public switched telephone network. It was first defined in 1988 in the CCITT red book.[1] Prior to ISDN, the telephone system was viewed as a way to transport voice, with some special services available for data. The key feature of ISDN is that it integrates speech and data on the same lines, adding features that were not available in the classic telephone system. The ISDN standards define several kinds of access interfaces, such as Basic Rate Interface (BRI), Primary Rate Interface (PRI), Narrowband ISDN (N-ISDN), and Broadband ISDN (B-ISDN). ISDN is a circuit-switched telephone network system, which also provides access to packet switched networks, designed to allow digital transmission of voice and data over ordinary telephone copper wires, resulting in potentially better voice quality than an analog phone can provide. It offers circuit-switched connections (for either voice or data), and packet-switched connections (for data), in increments of 64 kilobit/s. In some countries, ISDN found major market application for Internet access, in which ISDN typically provides a maximum of 128 kbit/s bandwidth in both upstream and downstream directions. Channel bonding can achieve a greater data rate; typically the ISDN B-channels of three or four BRIs (six to eight 64 kbit/s channels) are bonded.

Signaling System Number Seven (SS7) Associated Signaling (PRI), Non-Associated Signaling, Quasi-Associated (SS7) SSP (Service Switching Point or Signal Switching Point) - Switch STP (Signal Transfer Point) - Router SCP (Service Control Point) – Application access Point code SS7 like ISDN uses a form of CCS Signaling Modes While similar to ISDN-PRI, Signaling System Number Seven (SS7) uses different messaging for call setup and teardown. SS7 lets any SS7-enabled node to talk to any other, regardless of whether they have direct trunk connections between them.

Rtp scrambling Deep packet inspection (DPI) IP headers and Packet payload size/uniformity pTime

HA & load balancing DNS Multiple server Scalability (Horizontal, Vertical, Elastic) Proxy, SBC IP Float

Voip problems Fraud Timeout Echo & Delay NAT

LI - Lawful interception X1 interface X2 interface X3 interface X1 interface: This interface is known as Provisioning interface that is supported by Sonus EMS using SOAP XML/TCP. X2 interface: This interface is known as Signaling interface that is supported by Sonus SBC to send call data (signaling) messages over TCP using an optional IPSec tunnel. This interface encapsulates a copy of the SIP signaling message sent/received towards/from the target. X3 interface: This interface is known as Media interface that is supported by Sonus SBC to send call content (media) messages over UDP only. These media streams (audio/video/image/clearmode) carries a copy of the stream sent/received towards/from the target.

metric ASR ACD Latency Delay Jitter MOS

Web-Rtc WebRTC (Web Real-Time Communication) is a collection of communications protocols and application programming interfaces that enable real-time communication over peer-to-peer connections. This allows web browsers to not only request resources from backend servers, but also real-time information from browsers of other users.

references My friend, Wkipedia Introduction to SS7 Signaling – Patton.com Switching to VoIP – Ted Wallingford ETSI TR 101 943 V2.1.1 (2004-10) WebRTC (Web Real-Time Communication) is a collection of communications protocols and application programming interfaces that enable real-time communication over peer-to-peer connections. This allows web browsers to not only request resources from backend servers, but also real-time information from browsers of other users. This enables applications such as video conferencing, file transfer, chat, or desktop sharing without the need of either internal or external plugins.[1] WebRTC is being standardized by the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF). The reference implementation is released as free software under the terms of a BSD license. OpenWebRTC provides another free implementation based on the multimedia framework GStreamer.