Search completed in 0.92 seconds.
38 results for "negotiated":
Your results are loading. Please wait...
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). the default is false. see creating a data channel in using webrtc data channels for further information about this property. 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. the default is false. 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.ne...
...hannel() 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. candidate recommendation initial specification. ...
Index - Web APIs
Web › API › Index
xretransmits experimental, needsexample, property, rtcdatachannel, read-only, reference, webrtc, maxretransmits 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. 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). 3238 rtcdatachannel.onbufferedamountlow event handler, experimental, networking, property, rtcda...
...rting a new webrtc connection to a remote peer. 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. also 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. 3418 rtcpeerconnection.currentremotedescription api, media, property, rtcpeerconnection, read-only, refere...
...by the description was first instantiated. 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. also 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. 3419 rtcpeerconnection.defaulticeservers experimental, ice, property, rtcpeerconnection, reference, webrtc,...
...And 4 more matches
RTCPeerConnection - Web APIs
also 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 rtcsession...
...also 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.getdefaulticeservers() the getdefaulticeservers() method of the rtcpeerconnection interface returns an array of objects ba...
...motedescription returns a rtcsessiondescription describing the session (which includes configuration and media information) for the remote end of the connection. if this hasn't been set yet, this is null.sctp the read-only sctp property on the rtcpeerconnection interface returns an rtcsctptransport describing the sctp transport over which sctp data is being sent and received. 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. see signaling in lifetime of a webrtc session for more details ab...
...And 2 more matches
NSS 3.12.6 release notes
rver or client does not support safe renegotiation. the default setting for this option is false. 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. see bug 360421 for details. note: the tls server name indication for clients is already fully implemented in nss. new functions for sni (see ssl.
...ssl_sni_current_config_is_used: libssl must use the default cert and key. ssl_sni_send_alert: libssl must send the "unrecognized_name" alert. 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. cert_distnamesfromcertlist: generate an array of distinguish...
... cert_cacheocspresponsefromsidechannel: this function is intended for use when ocsp responses are provided via a side-channel, i.e. tls ocsp stapling (a.k.a. the status_request extension). ...
... bug 542538: nss: add function for recording ocsp stapled replies bug 544191: use system zlib on mac os x bug 544584: segmentation fault when enumerating the nss database bug 544586: various nss-sy...
Using WebRTC data channels - Web APIs
ransport and write your own code to signal to the other peer that it needs to connect to the new channel. let's look at each of these cases, starting with the first, which is the most common. automatic negotiation often, you can allow the peer connection to handle negotiating the rtcdatachannel connection for you. to do this, simply call createdatachannel() without specifying a value for the negotiated property, or specifying the property with a value of false. this will automatically trigger the rtcpeerconnection to handle the negotiations for you, causing the remote peer to create a data channel and linking the two together across the network. the rtcdatachannel object is returned immediately by createdatachannel(); you can tell when the connection has been made successfully by watching for ...
...t to be sent to the rtcdatachannel. 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 signals to the peer connection to not attempt to negotiate the channel on your behalf. then negotiate the connection out-of-band, using a web server or other means. 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. this will link the two objects across the rtcpeer...
... to negotiate the channel on your behalf. then negotiate the connection out-of-band, using a web server or other means. 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. this will link the two objects across the rtcpeerconnection. 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. doing this lets you create data channels ...
L20n Javascript API
e to negotiate against and freeze the context instance. when a context instance is frozen, no more resources can be added or linked. all io related to fetching the resource files takes place when a context instance freezes. when all resources have been fetched, parsed and compiled, the context instance will emit a ready event. the final list of locales supported by the context instance will be negotiated asynchronously by the negotiator registered by registerlocalenegotiator. ctx.requestlocales('pl'); ctx.requestlocales('fr-ca', 'fr'); if requestedlocales argument list is empty or undefined, the default locale from registerlocales will be used. ctx.requestlocales('pl'); if registerlocales hasn't been called, the special i-default locale is used, which means that the following minima...
...nce. var ctx = l20n.getcontext(); ctx.addresource('<hello "hello, world!">'); ctx.requestlocales(); requestlocales can be called multiple times after the context instance emitted the ready event, in order to change the current language fallback chain, for instance if requested by the user. 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. ctx.registerlocales('en-us', ['en-us', 'fr', 'pl']); ctx.requestlocales('fr-ca', 'fr'); ctx.ready(function() { // ctx.supportedlocales == ['fr']; }); ctx.addresource(text: string) add a string as the content of a resource to the context instance. the resource is added to all registered locales. ctx.a...
NSS 3.15.2 release notes
_3_15_2_rtm/src/ security advisories the following security-relevant bugs have been resolved in nss 3.15.2. users are encouraged to upgrade immediately. bug 894370 - (cve-2013-1739) avoid uninitialized data read in the event of a decryption failure. 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. specifically, the following cipher suites are now supported: tls_ecdhe_ecdsa_with_aes_128_gcm_sha256 tls_ecdhe_rsa_with_aes_128_gcm_sha256 tls_dhe_rsa_with_aes_128_gcm_sha256 tls_rsa_with_aes_128_gcm_sha256 new functions pk11_cipherfinal has been introduced, which is a simple alias for pk11_digestfinal. new types no new types have been introduced. new pkcs #...
...rt for aes-gcm ciphersuites that use the sha-256 prf bug 663313 - md2, md4, and md5 signatures are no longer accepted for ocsp or crls, consistent with their handling for general certificate signatures. 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. bug 848384 - deprecate the ssl cipher policy code, as it's no longer relevant. it is no longer necessary to call nss_setdomesticpolicy because all cipher suites are now allowed by default. a complete list of all bugs resolved in this release can be obtained at https://bugzilla.mozilla.org/buglist.cgi?resolution=fixed&classification=compon...
RTCDataChannel.id - Web APIs
Web › API › RTCDataChannel › id
each rtcpeerconnection can therefore have up to a theoretical maximum of 65,534 data channels on it, although the actual maximum may vary from browser to browser. in early versions of the webrtc specification, this property's name was stream. ...
... each rtcpeerconnection can therefore have up to a theoretical maximum of 65,534 data channels on it, although the actual maximum may vary from browser to browser. in early versions of the webrtc specification, this property's name was stream. ...
RTCDataChannel.stream - Web APIs
this property has been replaced with the rtcdatachannel.id property. ...
... this property has been replaced with the rtcdatachannel.id property. ...
RTCPeerConnection.createDataChannel() - Web APIs
transmit a message which fails the first time in unreliable mode. while this value is a16-bit unsigned number, each user agent may clamp it to whatever maximum it deems appropriate. default: null. protocol optional the name of the sub-protocol being used on the rtcdatachannel, if any; otherwise, the empty string (""). default: empty string, "". this string may not be longer than 65,535 bytes. 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. default: false. id optional an 16-bit numeric id for the channel; pe...
... specifications specification status comment webrtc 1.0: real-time communication between browsersthe definitio...
Writing WebSocket servers - Web APIs
the server may listen on any port it chooses, but if it chooses any port other than 80 or 443, it may have problems with firewalls and/or proxies. browsers generally require a secure connection for websockets, although they may offer an exception for local devices. the handshake is the "web" in websockets. it's the bridge from http to websockets. in the handshake, details of the connection are negotiated, and either party can back out before completion if the terms are unfavorable. the server must be careful to understand everything the client asks for, otherwise security issues can occur. tip: the request-uri (/chat here) has no defined meaning in the spec. so, many people use it to let one server handle multiple websocket applications. for example, example.com/chat could invoke a multiuser c...
...g handshake (detailed in section 5.5.1). upon receiving such a frame, the other peer sends a close frame in response. the first peer then closes the connection. any further data received after closing of connection is then discarded. miscellaneous websocket codes, extensions, subprotocols, etc. are registered at the iana websocket protocol registry. websocket extensions and subprotocols are negotiated via headers during the handshake. sometimes extensions and subprotocols very similar, but there is a clear distinction. extensions control the websocket frame and modify the payload, while subprotocols structure the websocket payload and never modify anything. extensions are optional and generalized (like compression); subprotocols are mandatory and localized (like ones for chat and for mmorpg ga...
Transport Layer Security - Web security
ent and the server, in order to prevent it from being read and interpreted by unauthorized parties. integrity tls ensures that between encrypting, transmitting, and decrypting the data, no information is lost, damaged, tampered with, or falsified. 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. ...
...mpered with, or falsified. 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. once parameters and a data exchange mode where application data, such http, is exchanged. cipher suites the primary parameters that the tls handshake negotiates is a cipher suite. 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. the cipher suite in tls 1.3 primarily governs the encryption of data, separate negotiation methods are used for key agreement and authentication. different software might use different ...
CSS3 - Archive of obsolete content
onal notation to describe the directionality of the image (rtl or ltr), allowing for bidi-sensitive images. the image-orientation property by adding the keyword from-image, allowing to follow exif data stored into images to be considered. adds: the image-set() functional notation to allow the definition to equivalent images at different resolution allowing for resolution-negotiated selection of images. the element() functional notation allowing the use of part of the page as image. the cross-fade() functional notation allowing to refer to intermediate images when transitioning between two images and defines the interpolation between two images. the conic-gradient() and repeating-conic-gradient() functional notation describing a new type of gradient. the ...
Integrated Authentication
f ntlmv2/lmv2. <i>this document is incomplete ...</i> flow diagram the diagram below shows how various components interact. configuration by default, mozilla rejects all spnego challenges from a web server. this is to protect the user from the possibility of dns-spoofing being used to stage a man-in-the-middle exploit (see bug 17578 for more info). moreover, with windows clients ntlm may be negotiated as the authentication protocol. so, it is paramount that the browser does not freely exchange ntlm user credentials with any server that requests them. the ntlm response includes a hash of the user's logon credentials. on older versions of windows this hash is computed using a relatively weak algorithm (see hertel for more info on ntlm authentication). mozilla currently supports a whitelist of s...
NSS 3.15 release notes
ocitem, which is now declared as obsolete. support for single-operation (eg: not multi-part) symmetric key encryption and decryption, via pk11_encrypt and pk11_decrypt. certutil has been updated to support creating name constraints extensions. 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. ssl_setstapledocspresponses - set's a stapled ocsp response for a tls server socket to return when clients send the status_request extension. in ocsp.h cert_postocsprequest - primarily intended for testing, permits the sending and receiving of raw ocsp request/responses. in secpkcs7.h sec_pkcs7verifydetachedsignatureatt...
NSS 3.20 release notes
as been added: tls_dhe_dss_with_aes_128_gcm_sha256 tls_dhe_dss_with_aes_128_cbc_sha256 tls_dhe_dss_with_aes_256_cbc_sha256 by default, the server side tls implementation will use dhe parameters with a size of 2048 bits when using dhe ciphersuites. 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. a new api ssl_dhegroupprefset has been added to nss, which allows a server application to select one or multiple of the embedded dhe parameters as the preferred parameters. the current implementation of nss will always use the first entry in the array that is passed as a parameter to the ssl_dhegroupprefset api. in future ver...
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
s as the limit is per browser and set to a very low number (6). the issue has been marked as "won't fix" in chrome and firefox. this limit is per browser + domain, so that means that you can open 6 sse connections across all of the tabs to www.example1.com and another 6 sse connections to www.example2.com. (from stackoverflow). when using http/2, the maximum number of simultaneous http streams is negotiated between the server and the client (defaults to 100). constructor eventsource() creates a new eventsource to handle receiving server-sent events from a specified url, optionally in credentials mode. properties this interface also inherits properties from its parent, eventtarget. eventsource.readystate read only a number representing the state of the connection. possible values are co...
MediaStream.onaddtrack - Web APIs
the mediastream.onaddtrack property is an eventhandler which specifies a function to be called when the addtrack event occurs on a mediastream instance. this happens when a new track of any kind is added to the media stream. 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. the addtrack event does not get fired when javascript code explicitly adds tracks to the stream (by calling addtrack()). syntax mediastream.onaddtrack = eventhandler; value this should be set to a function which you provide that accepts as input a mediastreamtrackevent object representing the addtrack event wh...
MediaStream.onremovetrack - Web APIs
the mediastream.onremovetrack property is an eventhandler which specifies a function to be called when the removetrack event occurs on a mediastream instance. this happens when a track of any kind is removed from the media stream. 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. the removetrack event does not get fired when javascript code explicitly removes tracks from the stream (by calling removetrack()). syntax mediastream.onremovetrack = eventhandler; value this should be set to a function which you provide that accepts as input a mediastreamtrackevent object represent...
RTCConfiguration.bundlePolicy - Web APIs
deo track and those two tracks are each assigned to the corresponding rtcdtlstransport. all other tracks are ignored by the connection. this is the default, and most compatible, policy. max-compat the ice agent intially creates one rtcdtlstransport for each media track and a separate one for the rtcdatachannel, if one is created. if the remote endpoint can't handle bundling, each media track is negotiated on its own separate transport. this introduces bundling but will fall back to not bundling if the remote peer can't handle it. max-bundle the ice agent starts by creating a single rtcdtlstransport to handle all of the connection's media. if the remote peer isn't bundle-compatible, only one media track is negotiated and the rest are ignored. this maximizes bundling at the risk of losing tracks i...
RTCConfiguration - Web APIs
ransport for each type of content added: audio, video, and data channels. if the remote endpoint is not bundle-aware, then each of these dtls transports then handles all the communication for one type of data. "max-compat" the ice agent initially creates one rtcdtlstransport per media track and a separate one for data channels. if the remote endpoint is not bundle-aware, everything is negotiated on these separate dtls transports. "max-bundle" the ice agent initially creates only a single rtcdtlstransport to carry all of the rtcpeerconnection's data. if the remote endpoint is not bundle-aware, then only a single track will be negotiated and the rest ignored. rtcicetransportpolicy enum the rtcicetransportpolicy enum defines string constants which can be used to limit th...
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
also 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. ...
RTCPeerConnection.currentRemoteDescription - Web APIs
also 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. ...
RTCPeerConnection.onnegotiationneeded - Web APIs
the rtcpeerconnection interface's onnegotiationneeded property is an eventlistener which specifies a function which is called to handle the negotiationneeded event when it occurs on an rtcpeerconnection instance. this event is fired when a change has occurred which requires session negotiation. this negotiation should be carried out as the offerer, because some session changes cannot be negotiated as the answerer. most commonly, the negotiationneeded event is fired after a send track is added to the rtcpeerconnection. if the session is modified in a manner that requires negotiation while a negotiation is already in progress, no negotiationneeded event will fire until negotiation completes, and only then if negotiation is still needed. syntax rtcpeerconnection.onnegotiationneeded = eventhandler; ...
RTCPeerConnection.removeTrack() - Web APIs
the rtcpeerconnection.removetrack() method tells the local end of the connection to stop sending media from the specified track, without actually removing the corresponding rtcrtpsender from the list of senders as reported by rtcpeerconnection.getsenders(). if the track is already stopped, or is not in the connection's senders list, this method has no effect. 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. a negotiationneeded event is sent to the rtcpeerconnection to let the local end know this negotiation must occur. syntax pc.removetrack(sender); parameters mediatrack a rtcrtpsender specifying the sender to remove from the connectio...
RTCPeerConnection.restartIce() - Web APIs
, once the offer is sent across your signaling mechanism and the remote peer has set its description as well. restartice() causes the negotiationneeded event to be fired on the rtcpeerconnection to inform the application that it should perform negotiation using its signaling channel. 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. the next time the connection's signalingstate changes to stable, the connection will fire the negotiationneeded event. this process continues until an ice restart has been successfully completed. syntax rtcpeerconnection.restartice(); parameters none. return value undefined. usage notes after calling restartice(), the n...
RTCPeerConnection.sctp - Web APIs
the sctp transport is used for transmitting and receiving data for any and all rtcdatachannels on the peer connection. syntax var sctp = rtcpeerconnection.sctp; value a rtcsctptransport object describing the sctp transport being used by the rtcpeerconnection for transmitting and receiving on its data channels, or null if sctp negotiation hasn't happened. ...
RTCPeerConnection.setRemoteDescription() - Web APIs
typeerror the specified rtcsessiondescriptioninit or rtcsessiondescription object is missing the type property, or no description parameter was provided at all. usage notes when you call setremotedescription(), the ice agent checks to make sure the rtcpeerconnection is in either the stable or have-remote-offer signalingstate. 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. in either of those two cases, we're at the beginning of the negotiation process, and the offer is set as the remote description. on the other hand, if we're in the middle of an ongoing negotiation and an offer is passed into setremotedescription(), the ice agent automatically ...
RTCRtpSender.replaceTrack() - Web APIs
nt one. the new track's frame rate is high enough to cause the codec's block rate to be exceeded. the new track is a video track and its raw or pre-encoded state differs from that of the original track. the new track is an audio track with a different number of channels fom the original. 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. examples switching video cameras // example to change video camera, suppose selected value saved into window.selectedcamera navigator.mediadevices .getusermedia({ video: { deviceid: { exact: window.selectedcamera } } }) .then(function(stream) { let videotrack = stream.getvideotracks()[0]; ...
RTCRtpTransceiver.mid - Web APIs
syntax var mediaid = rtcrtptransceiver.mid; value a domstring which uniquely identifies the pairing of source and destination of the transceiver's stream. its value is taken from the media id of the sdp m-line. this value is null if negotiation has not completed. ...
Using server-sent events - Web APIs
s, as the limit is per browser and is set to a very low number (6). the issue has been marked as "won't fix" in chrome and firefox. this limit is per browser + domain, which means that you can open 6 sse connections across all of the tabs to www.example1.com and another 6 sse connections to www.example2.com (per stackoverflow). when using http/2, the maximum number of simultaneous http streams is negotiated between the server and the client (defaults to 100). sending events from the server the server-side script that sends events needs to respond using the mime type text/event-stream. each notification is sent as a block of text terminated by a pair of newlines. for details on the format of the event stream, see event stream format. the php code for the example we're using here follows: date_de...
Lifetime of a WebRTC session - Web APIs
in step 4). descriptions, candidates, etc. -- more coming up ice restart sometimes, during the lifetime of a webrtc session, network conditions change. one of the users might transition from a cellular to a wifi network, or the network might become congested, for example. when this happens, the ice agent may choose to perform ice restart. 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. then media shifts to the new network connection and the old one is closed. note: different browsers support ice restart under different sets of conditions. not all browsers will perform ice restart due to network ...
WebSocket.extensions - Web APIs
syntax var extensions = awebsocket.extensions; value a domstring. ...
Writing a WebSocket server in C# - Web APIs
is a bitfield that breaks down as such: fin (bit 0) rsv1 (bit 1) rsv2 (bit 2) rsv3 (bit 3) opcode (bit 4:7) 1 0 0 0 0x1=0001 fin bit: this bit indicates whether the full message has been sent from the client. messages may be sent in frames, but for now we will keep things simple. rsv1, rsv2, rsv3: these bits must be 0 unless an extension is negotiated which supplies a nonzero value to them. opcode: these bits describe the type of message received. opcode 0x1 means this is a text message. full list of opcodes the second byte, which currently has a value of 131, is another bitfield that breaks down as such: mask (bit 0) payload length (bit 1:7) 1 0x83=0000011 mask bit: defines whether the "payload data" i...
Intl.DateTimeFormat() constructor - JavaScript
scribe the date-time components to use in formatted output, and their desired representations. 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. two algorithms are available for this negotiation and selected by the formatmatcher property: a fully specified "basic" algorithm and an implementation-dependent "best fit" algorithm. weekday the representation of the weekday. possible values are: "long" (e.g., thursday) "short" (e.g., thu) "narrow...