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RTCDataChannel.negotiated - Web APIs
the read-only rtcdatachannel property negotiated indicates whether the rtcdatachannel's connection was negotiated by the web app (true) or by the webrtc layer (false).
... syntax var negotiated = adatachannel.negotiated; value true if the rtcdatachannel's connection was negotiated by the web app itself; false if the negotiation was handled by the webrtc layer.
... example the code snippet below checks the value of negotiated; if it's true, a function called shutdownremotechannel() is called with the channel's id; presumably this would be implemented to transmit a shutdown signal to the remote peer prior to terminating the connection.
... if (datachannel.negotiated) { shutdownremotechannel(datachannel.id); } specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'rtcdatachannel.negotiated' in that specification.
Index - Web APIs
WebAPIIndex
3237 rtcdatachannel.negotiated networking, property, rtcdatachannel, read-only, reference, webrtc, webrtc api, data, datachannel, negotiated the read-only rtcdatachannel property negotiated indicates whether the rtcdatachannel's connection was negotiated by the web app (true) or by the webrtc layer (false).
... 3417 rtcpeerconnection.currentlocaldescription api, media, property, rtcpeerconnection, read-only, reference, sdp, webrtc, currentlocalconnection the read-only property rtcpeerconnection.currentlocaldescription returns an rtcsessiondescription object describing the local end of the connection as it was most recently successfully negotiated since the last time the rtcpeerconnection finished negotiating and connecting to a remote peer.
... 3418 rtcpeerconnection.currentremotedescription api, media, property, rtcpeerconnection, read-only, reference, sdp, webrtc, currentremotedescription the read-only property rtcpeerconnection.currentremotedescription returns an rtcsessiondescription object describing the remote end of the connection as it was most recently successfully negotiated since the last time the rtcpeerconnection finished negotiating and connecting to a remote peer.
...And 4 more matches
RTCPeerConnection - Web APIs
te property of the rtcpeerconnection interface indicates the current state of the peer connection by returning one of the string values specified by the enum rtcpeerconnectionstate.currentlocaldescription read only the read-only property rtcpeerconnection.currentlocaldescription returns an rtcsessiondescription object describing the local end of the connection as it was most recently successfully negotiated since the last time the rtcpeerconnection finished negotiating and connecting to a remote peer.
...so included is a list of any ice candidates that may already have been generated by the ice agent since the offer or answer represented by the description was first instantiated.currentremotedescription read only the read-only property rtcpeerconnection.currentremotedescription returns an rtcsessiondescription object describing the remote end of the connection as it was most recently successfully negotiated since the last time the rtcpeerconnection finished negotiating and connecting to a remote peer.
...if sctp hasn't been negotiated, this value is null.signalingstate read only the read-only signalingstate property on the rtcpeerconnection interface returns one of the string values specified by the rtcsignalingstate enum; these values describe the state of the signaling process on the local end of the connection while connecting or reconnecting to another peer.
...And 3 more matches
NSS 3.12.6 release notes
these options can also be set with the following ssl options: ssloptions.enablerenegotiation ssloptions.requiresafenegotiation new pseudo cipher suite value: tls_empty_renegotiation_info_scsv (cannot be negotiated) tls server name indication for servers tls server name indication (sni) for servers is almost fully implemented in nss 3.12.6.
... ssl_snisocketconfighook ssl_reconfigfd ssl_configserversessionidcachewithopt ssl_settrustanchors ssl_getnegotiatedhostinfo new enum for sni: sslsninametype (see sslt.h) new functions in cert.h certdistnames: duplicate distinguished name array.
... in ssl.h ssl_getimplementedciphers ssl_getnumimplementedciphers ssl_handshakenegotiatedextension new error codes in sslerr.h ssl_error_unsafe_negotiation ssl_error_rx_unexpected_uncompressed_record new types in sslt.h sslextensiontype ...
...g 536023: der_utctimetotime and der_generalizedtimetotime ignore all bytes after an embedded null bug 536474: add support for logging pre-master secrets bug 537356: implement new safe ssl3 & tls renegotiation bug 537795: nss_initcontext does not work with nss_registershutdown bug 537829: allow nss to build for android bug 540304: implement ssl_handshakenegotiatedextension bug 541228: remove an obsolete nspr version check in lib/util/secport.c bug 541231: nssinit.c doesn't need to include ssl.h and sslproto.h.
Using WebRTC data channels - Web APIs
to do this, simply call createdatachannel() without specifying a value for the negotiated property, or specifying the property with a value of false.
... let datachannel = pc.createdatachannel("myapp channel"); datachannel.addeventlistener("open", (event) => { begintransmission(datachannel); }); manual negotiation to manually negotiate the data channel connection, you need to first create a new rtcdatachannel object using the createdatachannel() method on the rtcpeerconnection, specifying in the options a negotiated property set to true.
...this process should signal to the remote peer that it should create its own rtcdatachannel with the negotiated property also set to true, using the same id.
... let datachannel = pc.createdatachannel("myapp channel", { negotiated: true }); datachannel.addeventlistener("open", (event) => { begintransmission(datachannel); }); requestremotechannel(datachannel.id); in this code snippet, the channel is created with negotiated set to true, then a function called requestremotechannel() is used to trigger negotiation, to create a remote channel with the same id as the local channel.
RTCPeerConnection.createDataChannel() - Web APIs
negotiated optional by default (false), data channels are negotiated in-band, where one side calls createdatachannel, and the other side listens to the rtcdatachannelevent event using the ondatachannel eventhandler .
... alternatively (true), they can be negotiated out of-band, where both sides call createdatachannel with an agreed-upon id.
...ction(event) { var channel = event.channel;  channel.onopen = function(event) { channel.send('hi back!'); } channel.onmessage = function(event) { console.log(event.data); } } alternatively, more symmetrical out-of-band negotiation can be used, using an agreed-upon id (0 here): // both sides var pc = new rtcpeerconnection(options); var channel = pc.createdatachannel("chat", {negotiated: true, id: 0}); channel.onopen = function(event) { channel.send('hi!'); } channel.onmessage = function(event) { console.log(event.data); } for a more thorough example showing how the connection and channel are established, see a simple rtcdatachannel sample.
L20n Javascript API
the final list of locales supported by the context instance will be negotiated asynchronously by the negotiator registered by registerlocalenegotiator.
... ctx.supportedlocales a read-only property which holds the current fallback chain of locales which was negotiated between all the available locales, the default locale and the user's preferred locales.
NSS 3.15.2 release notes
new in nss 3.15.2 new functionality aes-gcm ciphersuites: aes-gcm cipher suite (rfc 5288 and rfc 5289) support has been added when tls 1.2 is negotiated.
... bug 884178 - add pk11_cipherfinal macro bugs fixed in nss 3.15.2 bug 734007 - sizeof() used incorrectly bug 900971 - nssutil_readsecmoddb() leaks memory bug 681839 - allow ssl_handshakenegotiatedextension to be called before the handshake is finished.
RTCConfiguration.bundlePolicy - Web APIs
if the remote endpoint can't handle bundling, each media track is negotiated on its own separate transport.
...if the remote peer isn't bundle-compatible, only one media track is negotiated and the rest are ignored.
RTCConfiguration - Web APIs
if the remote endpoint is not bundle-aware, everything is negotiated on these separate dtls transports.
...if the remote endpoint is not bundle-aware, then only a single track will be negotiated and the rest ignored.
RTCRtpSender.replaceTrack() - Web APIs
media sources that have built-in encoders — such as hardware encoders — may not be able to provide the negotiated codec.
... software sources may not implement the negotiated codec.
Writing WebSocket servers - Web APIs
in the handshake, details of the connection are negotiated, and either party can back out before completion if the terms are unfavorable.
... websocket extensions and subprotocols are negotiated via headers during the handshake.
Transport Layer Security - Web security
a tls connection starts with a handshake phase where a client and server agree on a shared secret and important parameters, like cipher suites, are negotiated.
... in tls 1.2 and earlier, the negotiated cipher suite includes a set of cryptographic algorithms that together provide the negotiation of the shared secret, the means by which a server is authenticated, and the method that will be used to encrypt data.
CSS3 - Archive of obsolete content
adds: the image-set() functional notation to allow the definition to equivalent images at different resolution allowing for resolution-negotiated selection of images.
Integrated Authentication
moreover, with windows clients ntlm may be negotiated as the authentication protocol.
NSS 3.15 release notes
new functions in ssl.h ssl_peerstapledocspresponse - returns the server's stapled ocsp response, when used with a tls client socket that negotiated the status_request extension.
NSS 3.20 release notes
nss embeds fixed dhe parameters sized 2048, 3072, 4096, 6144 and 8192 bits, which were copied from version 08 of the internet-draft "negotiated finite field diffie-hellman ephemeral parameters for tls", appendix a.
NSS 3.21 release notes
plete dtls record in a udp packet ssl_error_no_supported_signature_algorithm - error code for when no valid signature and hash algorithm is available ssl_error_unsupported_signature_algorithm - error code for when an unsupported signature and hash algorithm is configured ssl_error_missing_extended_master_secret - error code for when the extended master secret is missing after having been negotiated ssl_error_unexpected_extended_master_secret - error code for receiving an extended master secret when previously not negotiated in sslt.h ssl_enable_extended_master_secret - configuration to enable the tls extended master secret extension (rfc 7627) ssl_preinfo_version - used with sslpreliminarychannelinfo to indicate that a tls version has been selected ssl_preinfo_cipher_su...
Python binding for NSS
ssl.nss_shutdown the following classes were added: sslciphersuiteinfo sslchannelinfo the following class methods were added: certificate.trust_flags certificate.set_trust_attributes sslsocket.set_ssl_version_range sslsocket.get_ssl_version_range sslsocket.get_ssl_channel_info sslsocket.get_negotiated_host sslsocket.connection_info_format_lines sslsocket.connection_info_format sslsocket.connection_info_str sslciphersuiteinfo.format_lines sslciphersuiteinfo.format sslchannelinfo.format_lines sslchannelinfo.format the following class properties were added: certificate.ssl_trust_flags certificate.email_trust_f...
EventSource - Web APIs
when using http/2, the maximum number of simultaneous http streams is negotiated between the server and the client (defaults to 100).
MediaStream.onaddtrack - Web APIs
this event is fired when the browser adds a track to the stream (such as when a rtcpeerconnection is renegotiated or a stream being captured using htmlmediaelement.capturestream() gets a new set of tracks because the media element being captured loaded a new source.
MediaStream.onremovetrack - Web APIs
this event is fired when the browser removes a track from the stream (such as when a rtcpeerconnection is renegotiated or a stream being captured using htmlmediaelement.capturestream() gets a new set of tracks because the media element being captured loaded a new source.
RTCDataChannel.id - Web APIs
WebAPIRTCDataChannelid
this id is set at the time the data channel is created, either by the user agent (if rtcdatachannel.negotiated is false) or by the site or app script (if negotiated is true).
RTCDataChannel.stream - Web APIs
this id is set at the time the data channel is created, either by the user agent (if rtcdatachannel.negotiated is false) or by the site or app script (if negotiated is true).
RTCDataChannel - Web APIs
lliseconds, the browser is allowed to take to attempt to transmit a message, as set when the data channel was created, or null.maxretransmits read only the read-only rtcdatachannel property maxretransmits returns the maximum number of times the browser should try to retransmit a message before giving up, as set when the data channel was created, or null, which indicates that there is no maximum.negotiated read only the read-only rtcdatachannel property negotiated indicates whether the rtcdatachannel's connection was negotiated by the web app (true) or by the webrtc layer (false).ordered read only the read-only rtcdatachannel property ordered indicates whether or not the data channel guarantees in-order delivery of messages; the default is true, which indicates that the data channel is indeed ord...
RTCPeerConnection.currentLocalDescription - Web APIs
the read-only property rtcpeerconnection.currentlocaldescription returns an rtcsessiondescription object describing the local end of the connection as it was most recently successfully negotiated since the last time the rtcpeerconnection finished negotiating and connecting to a remote peer.
RTCPeerConnection.currentRemoteDescription - Web APIs
the read-only property rtcpeerconnection.currentremotedescription returns an rtcsessiondescription object describing the remote end of the connection as it was most recently successfully negotiated since the last time the rtcpeerconnection finished negotiating and connecting to a remote peer.
RTCPeerConnection.onnegotiationneeded - Web APIs
this negotiation should be carried out as the offerer, because some session changes cannot be negotiated as the answerer.
RTCPeerConnection.removeTrack() - Web APIs
if the connection has already been negotiated (signalingstate is set to "stable"), it is marked as needing to be negotiated again; the remote peer won't experience the change until this negotiation occurs.
RTCPeerConnection.restartIce() - Web APIs
if negotiation fails to complete—either due to rollback or because incoming offers are in the process of being negotiated—the rtcpeerconnection will remember that you requested ice restart.
RTCPeerConnection.sctp - Web APIs
if sctp hasn't been negotiated, this value is null.
RTCPeerConnection.setRemoteDescription() - Web APIs
these states indicate that either an existing connection is being renegotiated or that an offer previously specified by an earlier call to setremotedescription() is to be replaced with the new offer.
RTCRtpTransceiver.mid - Web APIs
the read-only rtcrtptransceiver interface's mid property specifies the negotiated media id (mid) which the local and remote peers have agreed upon to uniquely identify the stream's pairing of sender and receiver.
Using server-sent events - Web APIs
when using http/2, the maximum number of simultaneous http streams is negotiated between the server and the client (defaults to 100).
Lifetime of a WebRTC session - Web APIs
this is a process by which the network connection is renegotiated, exactly the same way the initial ice negotiation is performed, with one exception: media continues to flow across the original network connection until the new one is up and running.
WebSocket.extensions - Web APIs
this is currently only the empty string or a list of extensions as negotiated by the connection.
Writing a WebSocket server in C# - Web APIs
rsv1, rsv2, rsv3: these bits must be 0 unless an extension is negotiated which supplies a nonzero value to them.
Intl.DateTimeFormat() constructor - JavaScript
implementations are required to support at least the following subsets: weekday, year, month, day, hour, minute, second weekday, year, month, day year, month, day year, month month, day hour, minute, second hour, minute implementations may support other subsets, and requests will be negotiated against all available subset-representation combinations to find the best match.