Tag Archives: Real-time Transport Protocol

WebRTC layers

WebRTC stands for Web Real-Time Communications and  introduces  a real-time media framework in the browser core alongside associated JavaScript APIs for controlling the media frame and HTML5 tags for displaying.

From a technical point of view, WebRTC will hide all the complexity of real-time media behind a very simple JavaScript API . 

WebRTC simplified :

In simple words its a phenomenal thing , that will revolutionize internet telephony .  Also it will emerge to be platform independent ( ie any browser , any desktop operating system any mobile Operating system ) .

WebRTC allows anybody to introduce real-time communication to their web page as simply as introducing a table.

Codec Confusion :

Audiio Codecs

Currently VP8 is the codec of choice since it is royalty free. In mobility today, the codec of choice is h264. H264 is not royalty free. But it is native in most mobile handsets due to its high performance.

Voice Codecs

Opus is a lossy audio compression format developed by the Internet Engineering Task Force (IETF) targeting a broad range of interactive real-time applications over the Internet, from speech to music. As an open format standardized through RFC
6716, a reference implementation is provided under the 3-clause BSD license. All known software patents Which cover Opus are licensed under royalty-free terms.
G.711 is an ITU (International Telecommunications Union) standard for  audio compression. It is primarily used in telephony. The standard was released in 1972. It is the required standard in many voice-based systems  and technologies, for example in H.320 and H.323 specifications.
Speex is a patent-free audio compression format designed for speech and also  a free software speech codec that is used in VoIP applications and podcasts. Some consider Speex obsolete, with Opus as its official successor, but since
significant content is out there using Speex, it will not disappear anytime soon.
G.722 is an ITU standard 7 kHz Wideband audio codec operating at 48, 56 and 64 kbit/s. It was approved by ITU-T in 1988. G722 provides improved speech quality due to a wider speech bandwidth of up to 50-7000 Hz compared to G.711 of 300–3400 Hz.

AMR-WB Adaptive Multi-rate Wideband is a patented wideband speech coding standard that provides improved speech quality due to a wider speech bandwidth of 50–7000 Hz. Its data rate is between 6-12 kbit/s, and the codec is generally available on mobile phones.

Architecture :

WebRTC offers web application developers the ability to write rich, realtime multimedia applications (think video chat) on the web, without requiring plugins, downloads or installs. It’s purpose is to help build a strong RTC platform that works across multiple web browsers, across multiple platforms.

WebRTCpublicdiagramforwebsite

Web API – An API to be used by third party developers for developing web based videochat-like applications.

WebRTC Native C++ API – An API layer that enables browser makers to easily implement the Web API proposal.

Transport / Session

The session components are built by re-using components from libjingle, without using or requiring the xmpp/jingle protocol.

RTP Stack – A network stack for RTP, the Real Time Protocol.

STUN/ICE – A component allowing calls to use the STUN and ICE mechanisms to establish connections across various types of networks.

Session Management

An abstracted session layer, allowing for call setup and management layer. This leaves the protocol implementation decision to the application developer.

VoiceEngine

VoiceEngine is a framework for the audio media chain, from sound card to the network.

iSAC / iLBC / Opus

iSAC: A wideband and super wideband audio codec for VoIP and streaming audio. iSAC uses 16 kHz or 32 kHz sampling frequency with an adaptive and variable bit rate of 12 to 52 kbps.

iLBC: A narrowband speech codec for VoIP and streaming audio. Uses 8 kHz sampling frequency with a bitrate of 15.2 kbps for 20ms frames and 13.33 kbps for 30ms frames. Defined by IETF RFCs 3951 and 3952.

Opus: Supports constant and variable bitrate encoding from 6 kbit/s to 510 kbit/s, frame sizes from 2.5 ms to 60 ms, and various sampling rates from 8 kHz (with 4 kHz bandwidth) to 48 kHz (with 20 kHz bandwidth, where the entire hearing range of the human auditory system can be reproduced). Defined by IETF RFC 6176.

NetEQ for Voice– A dynamic jitter buffer and error concealment algorithm used for concealing the negative effects of network jitter and packet loss. Keeps latency as low as possible while maintaining the highest voice quality.

Acoustic Echo Canceler (AEC) – The Acoustic Echo Canceler is a software based signal processing component that removes, in real time, the acoustic echo resulting from the voice being played out coming into the active microphone.

Noise Reduction (NR) -The Noise Reduction component is a software based signal processing component that removes certain types of background noise usually associated with VoIP. (Hiss, fan noise, etc…)

VideoEngine 

VideoEngine is a framework video media chain for video, from camera to the network, and from network to the screen.

VP8 –Video codec from the WebM Project. Well suited for RTC as it is designed for low latency.
Video Jitter Buffer – Dynamic Jitter Buffer for video. Helps conceal the effects of jitter and packet loss on overall video quality.
Image enhancements -For example, removes video noise from the image capture by the webcam.


w3c

—Media Stream Functions

—API for connecting processing functions to media devices and network connections, including media manipulation functions.

—Audio Stream Functions

—An extension of the Media Stream Functions to process audio streams (e.g. automatic gain control, mute functions and echo cancellation).

—Video Stream Functions

—An extension of the Media Stream Functions to process video streams (e.g. bandwidth limiting, image manipulation or “video mute“).

Functional Component Functions

—API to query presence of WebRTC components in an implementation, instantiate them, and connect them to media streams.

—P2P Connection Functions

—API functions to support establishing signalling protocol agnostic peer-to-peer connections between Web browsers

  • API specification Availability

WebRTC 1.0: Real-time Communication Between Browsers –  Draft 3 June 2013 available

  • -Implementation Library: WebRTC Native APIs

Media Capture and Streams – Draft 16 May 2013

  • Supported by Chrome , Firefox , Opera in desktop of all OS ( Linux , Windows , Mac )
  • Supported by Chrome , Firefox  in Mobile browsers ( android )

ietf

  • —Communication model
  • —Security model
  • —Firewall and NAT traversal
  • —Media functions
  • —Functionality such as media codecs, security algorithms, etc.,
  • —Media formats
  • —Transport of non media data between clients
  • —Input to W3C for APIs development
  • Interworking with legacy VoIP equipment

WG RFC   Date

  • draft-ietf-rtcweb-audio-02      2013-08-02
  • draft-ietf-rtcweb-data-channel-05      2013-07-15
  • draft-ietf-rtcweb-data-protocol-00      2013-07-15
  • draft-ietf-rtcweb-jsep-03      2013-02-27
  • draft-ietf-rtcweb-overview-07      2013-08-14
  • draft-ietf-rtcweb-rtp-usage-07     2013-07-15
  • draft-ietf-rtcweb-security-05      2013-07-15
  • draft-ietf-rtcweb-security-arch-07      2013-07-15
  • draft-ietf-rtcweb-transports-00      2013-08-19
  • draft-ietf-rtcweb-use-cases-and-reqs-11      2013-06-27
  • Plus over 20 discussion RFC drafts

Next -> webRTC business benefits


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IP Multimedia Subsystem ( IMS )

 IMS is a an architectural framework for IP based multimedia rich communications. It was standardized by a group called 3GPP formed in 1999.
It started as an enabler for 3rd generation mobile networks in European market and later spread to wirelne networks too . IMS became the key to  Fixed Mobile Convergence (FMC).
Based on IETF Protocols (such as SIP, RTP, RTSP, COPS, DIAMETER, etc.) , IMS is  now crucial for controlling conmmunication in a IP based Next Genration Network ( NGN ).
Communication service providers and telecom operators are migrating from circuit-switched networks to IMS technology with the increasing bandwidth (5G) and user expectations.
ims layers

Why IMS ?

Early days TDM networks were not robust enough to support emerging technologies and data networking. There was a need to migrate from voic eonly network to Triple play network ( voice , video and data ).

Other factors included :

  • rapid service development
  • service availiability in both home and roaming network
  • wireline and wireless convergence

Due to these above mentioned reasons TDM was outdated and IMS gained support .

 

What benefits does IMS bring ?

It offers counteless applications around rich multimedia services on wireless , packet swtched and even tradional circuit switched networks.

Easier to Create and Deploy New Applications and Services
  •  Enhanced applications are easier to develop due to open APIs and common network services.
  • Third-party developers can offer their own applications and use common network services, sharing profits with minimal risk
  •  New services involving concurrent sessions of multimedia (voice, video, and data) during the same call are now possible
  • Reduced time-to-market for new services is possible because service providers are not tied to the timescales and functions of their primary NEPs
Capture New Subscribers, Retain Current Subscribers
  • Better voice quality for business applications, such as conferencing, is possible
  • Wireless applications (like SMS, and so on) can be offered to wire line or broadband subscribers.
  • Service providers can more easily offer bundled services.
Lower Operating and Capital Costs
  • Cost-effective implementation of services is possible across multiple transports, such as Push-To-Talk (PTT), presence and Location-Based Services (LBS), Fixed-Mobile Convergence (FMC), mobile video services, and so on.
  • Common provisioning, management, and billing systems are supported for all networks.
  • Significantly lower transport costs result when moving from time-switched to packet-switched channels.
  • Service providers can take advantage of competitive offerings from multiple NEPs for most network elements.
  • IMS results in reduced expenses for delivering licensed content to subscribers of different types of devices, encodings, or networks.

 

The reason for widespread adoption of IMS is also that it follows standards and open interfaces  from 3GPP and ETSI, also is flexible for policy control , OSS/BSS , Value Added Services etc .

 

IMS features

1. Abstraction from Underlying Network :
IMS is essentially leading towards an open and standardized network and interface ,  irrespective of underlay network.
2. Fixed /Mobile Convergence 
Inter operability with Circuit Switched (CS) Mobile application Part (MAP)
3. Roaming 
Location awareness between home and visiting network.
4. Application layer Call Control
IMS application layer has the provision for defining proxy or B2BUA based call flow completion . This leads to operator being able to introduce business logic into call sessions.
IMS is supplemented by SIP (IETF ) , Diameter ( IETF) and H248(ITU-T). The release cycle of IMS is as follows :
  • 2002-03-14 Rel-5  : IMS was introduced with SIP. Qos voice over MGW.
  • 2004-12-16 Rel-6 : Services like emergency , voice call continuity , IPCAN ( IP connectivity Access Network )
  • 2005-09-28 Rel-7 : Single Radio Voice Call Continuity , multimedia telephony,eCall ,ICS
  • 2008-12-11 Rel-8 : IMS centralized services , supplementary services and internetworking between  IMS and  Circuit Switched Networks,charging , QoS
  • 2009-12-10 Rel-9 : IMS emergency numbers on GPRS , EPS(Enhanced packet system) , Custom alert tone , MM broadcast/Multicast
  • 2011-3-23 Rel-10 : home NodeB, M2M, Roaming and Inter UE transfer
  • 2012-09-12 Rel-11 :-tbd
  • 2014-09-17 Rel-12 :- tbd
  • 2015-12-11 Rel-13 :- tbd

IMS Layers

Majorly IMS is divided into 3 horizontal layers given below :

2014-05-24_0015

•Transport / MediaEndpoint Layer

Unifies transports and media from analog, digital, or broadband formats to Real-time Transport Protocol (RTP) and SIP protocols. This is accomplished by media gateways and signaling gateways.

It also includes media servers with media processing elements to allow for announcements, in-band signaling, and conferencing. These media servers are shared across all applications (voicemail, interactive response systems, push-to-talk, and so on), maximizing statistical use of the equipment and creating a common base of media services without “hard-coding” these services into the applications.

•Session and Control Layer

This layer arranges logical connections between various other network elements. It provides registration of end-points, routing of SIP messages, and overall coordination of media and signaling resources.

IMS core which is part of this layer primarily contains 2 important elements Call Session Control Function (CSCF) and Home Subscriber Server (HSS) database. These are explained below 

HSS Home Subscriber Server

It is a database of user profiles and location information . It is responsible for name/address resolution and also authorization/authentication .

CSCF Call Session Control Function

Handles most routing, session and security related operation for SIP messages . It is further divided into 3 parts :

  • Proxy CSCF: P_CSCF is the first point of contact from any SIP UA. It proxies UE requests to subsystem.
  • Serving CSCF: S-CSCF is a powerful part of IMS Core as it decides how UE request will be forwarded to the application servers.
  • Interrogating CSCF: I-CSCF initiates the assignment of a user to an S-CSCF (by querying the HSS) during registration.

•Application Services Layer

 The Application Services Layer contains multiple Application Servers (AS), such as:
  • Telephony Application Server (TAS) – for defining custom call flow logic
  • IP Multimedia Services Switching Function (IM-SSF)
  • Open Service Access Gateway (OSA-GW), and so on.

Additional Links :


Update on IMS :

IMS has been mandated as the control architecture for Voice over LTE (VoLTE) networks. Also IMS is being widely adopted to mange traffic for Voice over WiFi (VoWiFi) systems.