Release Notes
Release 7.1
E55133-03
July 2015
These release notes list the documentation updates since General Availability (GA), new features and known issues for WebRTC Session Controller.
Table 1 lists the changes made to this documentation set since the original release date.
Table 1 WebRTC Session Controller Documentation Change Log
Date | Details |
---|---|
July 2015 |
|
April 2015 |
|
March 2015 |
Original release. |
This section briefly describes the new features of WebRTC Session Controller.
WebRTC Session Controller's JSON protocol, based upon the Session Description Protocol (SDP) offer-answer model, supports the following new extensibility scenario:
Arbitrary Representational State Transformation (REST) calls from Groovy scripts to REST endpoints
For more information, see "Initiating REST Calls from Groovy" in Oracle Communications WebRTC Session Controller Extension Developer's Guide.
WebRTC Session Controller supplies a comprehensive JavaScript Development Kit that provides a framework for creating and extending WebRTC applications, while shielding Web developers from much of the complexity of WebRTC application development.
The JavaScript library provides a set of Application Programming Interface (API) packages that handle the following new tasks:
Capabilities exchange
Stand alone messaging
One on one and group chat
File transfer
The JavaScript API maps to the underlying JSON-base protocol, and is completely extensible. In addition, the API is fully compatible with Firefox, Chrome, and Opera Web browsers on any of their supported platforms.
For more information on the WebRTC Session Controller JavaScript API, see Oracle Communications WebRTC Session Controller Application Developer's Guide.
The WebRTC Session Controller Android and iOS SDKs enable you to integrate your Android and iOS applications with core WebRTC Session Controller functions. You can use the Android and iOS SDKs to implement the following features:
Audio calls between an Android or iOS application and any other WebRTC-enabled application, a Session Initialization Protocol (SIP) endpoint, or a Public Switched Telephone Network endpoint using a SIP trunk.
Video calls between an Android or iOS application and any other WebRTC-enabled application, with suitable video conferencing support.
Seamless upgrading of an audio call to a video call and downgrading of a video call to an audio call.
Support for Interactive Connectivity Establishment (ICE) server configuration, including support for Trickle ICE.
Transparent session reconnection following network connectivity interruption.
For more information on the Android SDK, see "Developing WebRTC-enabled Android Applications" in Oracle Communications WebRTC Session Controller Application Developer's Guide. For more information on the iOS SDK, see "Developing WebRTC-enabled iOS Applications" in Oracle Communications WebRTC Session Controller Application Developer's Guide.
The Lightweight Proxy Registrar introduces a layer between the WebRTC Session Controller Signaling Engine and the Proxy Registrar. The Lightweight Proxy Registrar reduces resources consumed in the Proxy Registrar, which reduces overall cost.
WebRTC-based clients come and go as people open and close their browsers. Each WebSocket connection to WebRTC Session Controller triggers a SIP registration. A single user often has multiple devices, which equates to multiple endpoints and requires more register and unregister requests. These factors make the number of registrations at any time difficult to predict, making it harder to plan needed resources. Any cost associated with registration might not be well known and could be problematic.
The Lightweight Proxy Register addresses these problems in one of two ways:
Multiplexing registration requests from many WebRTC endpoints into a single SIP registration per user
Managing all registrations, leaving no registrations for the Proxy Registrar. In this case, an external system must route inbound calls to WebRTC Session Controller, for example, by using static routes based on the domain.
Customers who do not want or need a SIP or IMS integration do not need to use them This case is suitable for an enterprise that only wants to connect WebRTC endpoints. For more information on the Lightweight Proxy Registrar, see "Using the Lightweight Proxy Registrar" in Oracle Communications WebRTC Session Controller Administrator's Guide.
Signaling Engine now collects data in an event detail record (EDR) for each event that occurs in a subsession for the Call, Chat, and File Transfer packages. Each engine creates event detail records and writes them to a file, oracle.wsc.core.edr, for the sessions that it owns.
For more information, see "Accessing Event Detail Records" in Oracle Communications WebRTC Session Controller System Administrator's Guide.
To improve performance when Web applications negotiate connections from private networks behind Network Address Translation (NAT)-enabled routers using the Interactive Connectivity Establishment (ICE) protocol, WebRTC Session Controller supports the draft IETF Trickle ICE specification: http://tools.ietf.org/html/draft-ietf-mmusic-trickle-ice-01
.
For more information, see "Enabling Trickle ICE to Improve Application Performance" in Oracle Communications WebRTC Session Controller Application Developer's Guide.
The new features in WebRTC Session Controller Media Engine (Media Engine) include:
Message Session Relay Protocol (MSRP) Interworking
Configuring Static DTLS Certificates
Call Management Enhancements
Trickle ICE Support
TURN Over TCP/TLS
Certified Platforms for the Media Engine
Data Channel Support
TURN Implementation Updates
The Media Engine now supports Message Session Relay Protocol (MSRP) interworking.
MSRP interworking allows communication between WebRTC and Rich Communication Suite (RCS) endpoints. This protocol is used for transmitting a series of instant message chats and file transfers within the context of a session.
For more information on MSRP, see https://tools.ietf.org/html/rfc4975
.
The Media Engine supports Datagram Transport Layer Security in WebRTC environments. When you implement DTLS, you must configure a static certificate via the default-dtls-settings configuration object.
The Media Engine fully supports trickle ICE, a draft extension to RFC 5245 that allows ICE agents to incrementally exchange remote candidate information. Trickle ICE support considerably reduces call setup time by allowing ICE to run before the candidate harvesting phase has completed by sending empty or partial media candidate lists in the SDP.
Note:
Trickle ICE is not supported in augmented ICE scenarios. If both trickle ICE and augmented ICE are configured, disable augmented ICE to achieve the best possible performance.The Media Engine supports Traversal Using Relay NAT (TURN) over Transmission Control Protocol (TCP) for WebRTC. The TURN protocol assists clients located behind NAT devices to reach peers. In cases where clients and peers cannot create a direct communication path (for example, if both endpoints are behind individual NATs), it is necessary for an intermediate network device to relay data. The Media Engine TURN Server acts as a communication-enabling alternative for such cases, relaying data between the NAT-hidden clients. When used with ICE, the Media Engine TURN Server relay transport addresses are included in SDP ICE candidates received from clients. For more information on TURN, visit http://tools.ietf.org/search/rfc5766
.
Several platforms have been tested and certified for use with the Media Engine.
The following platforms have been certified for use with the Media Engine:
Sun Netra X3-2
HPDL360 G7
HPDL585 G7
HPDL320 G8
HPDL360 G8
Cisco C200
The following VM platforms have been certified for use with the Media Engine:
OVM 3.2.8
VMware ESXi 5.5
XEN 3.4.3
The Media Engine now supports data channels for anchored media.
Data channels use the SCTP protocol as a generic transport service which allows web browsers to exchange non-media data between peers.
For more information on data channels, see https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-11
.
Note:
You must have media anchored for data channels to work properly.This section provides a list of the new configuration objects in this release.
in-msrp-session-leg
out-msrp-session-leg
This section provides a summary of the new configuration properties added to this release.
This section describes changes that have been applied to the Management Information Base (MIB) object definitions.
The following table lists the changed MIB tables.
MIB Object Name | Description |
---|---|
activeMsrpSessions |
|
iceCandidatePairStatus |
|
licenseDetails |
|
msrpConnections |
|
processes |
|
systemInfo |
|
Table 4 lists the known issues in this release.
Bug ID | Description |
---|---|
19761420 |
Issue The USB install does not recognize the RAID array and cannot be used to install the code. Workaround To mitigate this issue, turn the USB stick into a rescue stick and the installation succeeds. |
20019513 |
Issue A SIP fault occurs when you use the call-control-custom command and your configuration includes the request-uri-specification object, Workaround Leave the request-uri-specification object unconfigured if you must use the call-control-custom command. |
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.
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Oracle Communications WebRTC Session Controller Release Notes, Release 7.1
E55133-03
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