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WebRTC School Qualified Developer - WSQD » The SIP School Translation Project

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WebRTC School Qualified Developer - WSQD

WebRTC includes a standardized signaling protocol
  • Or provide a new one...
FALSE
  • Or provide a new one...
TRUE
  • Or provide a new one...
DTLS and TLS are examples of ______ protocols used by WebRTC
  • Or provide a new one...
NAT traversal
  • Or provide a new one...
security
  • Or provide a new one...
signaling
  • Or provide a new one...
media
  • Or provide a new one...
Which protocol is used to describe WebRTC media sessions?
  • Or provide a new one...
SDP
  • Or provide a new one...
TCP
  • Or provide a new one...
HTTP
  • Or provide a new one...
SIP
  • Or provide a new one...
For media transport, WebRTC uses the secure profile of what protocol?
  • Or provide a new one...
SIP
  • Or provide a new one...
TCP
  • Or provide a new one...
HTTP
  • Or provide a new one...
RTP
  • Or provide a new one...
Support of IPv6 is a major problem for WebRTC
  • Or provide a new one...
FALSE
  • Or provide a new one...
TRUE
  • Or provide a new one...
Which are the two other protocols ICE relies on?
  • Or provide a new one...
TURN
  • Or provide a new one...
HTTP
  • Or provide a new one...
TLS
  • Or provide a new one...
STUN
  • Or provide a new one...
TCP
  • Or provide a new one...
NAT stands for
  • Or provide a new one...
Network Awarness Topology
  • Or provide a new one...
Network Address Translator
  • Or provide a new one...
Network Address Transport
  • Or provide a new one...
aNnoying Addressable Turncoats
  • Or provide a new one...
WebRTC's peer-to-peer media is important because:
  • Or provide a new one...
privacy
  • Or provide a new one...
low latency
  • Or provide a new one...
security
  • Or provide a new one...
All of above
  • Or provide a new one...
Which are the main standards bodies involved in WebRTC?
  • Or provide a new one...
IETF
  • Or provide a new one...
SIP Forum
  • Or provide a new one...
IEEE
  • Or provide a new one...
ITU-T
  • Or provide a new one...
W3C
  • Or provide a new one...
IMTC
  • Or provide a new one...
WebRTC media sessions are always encrypted
  • Or provide a new one...
FALSE
  • Or provide a new one...
TRUE
  • Or provide a new one...
A variety of codecs can be downloaded and used by WebRTC users
  • Or provide a new one...
FALSE
  • Or provide a new one...
TRUE
  • Or provide a new one...
WebRTC includes a bulit-in NAT traversal solution allowing peer-to-peer media in many situations
  • Or provide a new one...
FALSE
  • Or provide a new one...
TRUE
  • Or provide a new one...
The real-time communications media session in WebRTC is known as a:
  • Or provide a new one...
Data Channel
  • Or provide a new one...
ICE Connection
  • Or provide a new one...
Signaling Channel
  • Or provide a new one...
Peer Connection
  • Or provide a new one...
WebRTC allows real-time communication without a plugin or install
  • Or provide a new one...
FALSE
  • Or provide a new one...
TRUE
  • Or provide a new one...
ICE provides two essential functions to WebRTC:
  • Or provide a new one...
Authentication and Privacy
  • Or provide a new one...
NAT traversal and communication consent
  • Or provide a new one...
NAT traversal and encryption
  • Or provide a new one...
Synchronization and Mixing
  • Or provide a new one...
Three common signaling transports for WebRTC are:
  • Or provide a new one...
WebSocket, HTTP and data channel
  • Or provide a new one...
TLS, DTLS and SRTP
  • Or provide a new one...
SIP, H.323 and MGCP
  • Or provide a new one...
WebSocket, Secure WebSocket and TCP
  • Or provide a new one...
A WebSocket can be opened directly between two browsers.
  • Or provide a new one...
FALSE
  • Or provide a new one...
TRUE
  • Or provide a new one...
A data channel can be opened directly between two browsers.
  • Or provide a new one...
FALSE
  • Or provide a new one...
TRUE
  • Or provide a new one...
Data channel signaling eliminates the need for any type of signaling server.
  • Or provide a new one...
FALSE
  • Or provide a new one...
TRUE
  • Or provide a new one...
HTTP polling and WebSocket proxy are two examples of WebRTC ________.
  • Or provide a new one...
signaling
  • Or provide a new one...
media transport
  • Or provide a new one...
HTML5
  • Or provide a new one...
security
  • Or provide a new one...
The difference between WebSocket and Secure WebSocket is _____ transport.
  • Or provide a new one...
TLS
  • Or provide a new one...
UDP
  • Or provide a new one...
TURN
  • Or provide a new one...
SRTP
  • Or provide a new one...
Data channel signaling provides:
  • Or provide a new one...
privacy for signaling
  • Or provide a new one...
minimal load on signaling server
  • Or provide a new one...
low latency signaling
  • Or provide a new one...
All of above
  • Or provide a new one...
WebRTC introduces new possible attacks on browsers.
  • Or provide a new one...
FALSE
  • Or provide a new one...
TRUE
  • Or provide a new one...
TLS uses digital certificates to authenticate web servers.
  • Or provide a new one...
FALSE
  • Or provide a new one...
TRUE
  • Or provide a new one...
Key management for SRTP is:
  • Or provide a new one...
the way in which symmetric keys are generated/exchanged for a media session.
  • Or provide a new one...
optional
  • Or provide a new one...
the authentication of parties in communication
  • Or provide a new one...
the secure storage of SRTP keys
  • Or provide a new one...
DTLS-SRTP key agreement is authenticated using fingerprints.
  • Or provide a new one...
FALSE
  • Or provide a new one...
TRUE
  • Or provide a new one...
Identity proxy is:
  • Or provide a new one...
a way of securing web browsing
  • Or provide a new one...
a signaling authentication mechanism for WebRTC
  • Or provide a new one...
a new identity scheme for WebRTC that uses DTLS-SRTP.
  • Or provide a new one...
a SIP server that provides identity information
  • Or provide a new one...
Existing web identity services could be utilized by the Identity Proxy mechanism.
  • Or provide a new one...
FALSE
  • Or provide a new one...
TRUE
  • Or provide a new one...
Ensuring that browsers only send media packets to browsers that are expecting and wish to receive media is known as ___________.
  • Or provide a new one...
user permission grant
  • Or provide a new one...
trust
  • Or provide a new one...
security
  • Or provide a new one...
communication consent
  • Or provide a new one...
Every 'major' browser supports WebSockets.
  • Or provide a new one...
FALSE
  • Or provide a new one...
TRUE
  • Or provide a new one...
A device that maps an 'inside' IP address to an 'outside' IP address is known as a _______.
  • Or provide a new one...
DIRECTORY
  • Or provide a new one...
REGISTRAR
  • Or provide a new one...
FIREWALL
  • Or provide a new one...
PROXY
  • Or provide a new one...
NAT
  • Or provide a new one...
For NAT traversal a WebRTC 'enabled' web browser uses a protocol known as ____
  • Or provide a new one...
standard-ICE
  • Or provide a new one...
Fast-ICE
  • Or provide a new one...
ICE
  • Or provide a new one...
ICERTC
  • Or provide a new one...
ICE-Lite
  • Or provide a new one...
Standards documents published by the IETF are known as ____
  • Or provide a new one...
RFCs
  • Or provide a new one...
schema
  • Or provide a new one...
blueprints
  • Or provide a new one...
policy
  • Or provide a new one...
WebSockets can be used by WebRTC for the _______ channel.
  • Or provide a new one...
signaling
  • Or provide a new one...
video
  • Or provide a new one...
audio
  • Or provide a new one...
Sending and receiving ICE connectivity checks is also known as hole _________.
  • Or provide a new one...
drilling
  • Or provide a new one...
opening
  • Or provide a new one...
creating
  • Or provide a new one...
punching
  • Or provide a new one...
A version of ICE that shortens ICE processing time is known as _____ ICE.
  • Or provide a new one...
trickle
  • Or provide a new one...
fast
  • Or provide a new one...
secure
  • Or provide a new one...
candidate
  • Or provide a new one...
channelized
  • Or provide a new one...
IP address privacy is possible in WebRTC if a ______ server is used for media.
  • Or provide a new one...
PROXY
  • Or provide a new one...
STUN
  • Or provide a new one...
WEBSOCKET
  • Or provide a new one...
TURN
  • Or provide a new one...
The 'C' in WebRTC stands for _________
  • Or provide a new one...
Collaboration
  • Or provide a new one...
Contact
  • Or provide a new one...
Connection
  • Or provide a new one...
Communication
  • Or provide a new one...
Not every 'major' browser supports WebRTC today.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
The WebRTC JavaScript APIs are defined by the ______ standards organization.
  • Or provide a new one...
IETF
  • Or provide a new one...
W3C
  • Or provide a new one...
Internet2
  • Or provide a new one...
ITU
  • Or provide a new one...
The Media Capture and Streams specification defines the Peer Connection APIs
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
Recommendation is the final stage in the W3C standards process
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
All media flows in WebRTC are bidirectional
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
WebRTC supports sending 'how many' audio and video flows?
  • Or provide a new one...
At most one
  • Or provide a new one...
Exactly one
  • Or provide a new one...
More than one
  • Or provide a new one...
A media source in WebRTC is _________
  • Or provide a new one...
What gets sent over a Peer Connection
  • Or provide a new one...
What you use to configure devices
  • Or provide a new one...
How a local device is represented
  • Or provide a new one...
All of the above
  • Or provide a new one...
A MediaStreamTrack can be associated with multiple input sources
  • Or provide a new one...
FALSE
  • Or provide a new one...
TRUE
  • Or provide a new one...
A MediaStreamTrack cannot simultaneously contain both audio and video.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
A MediaStreamTrack can have ______
  • Or provide a new one...
Both an id and a label
  • Or provide a new one...
An id but no label
  • Or provide a new one...
A label but no id
  • Or provide a new one...
Neither an id nor a label
  • Or provide a new one...
The label attribute for a MediaStreamTrack is generated by the browser.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
A MediaStreamTrack can be muted by ________
  • Or provide a new one...
The application
  • Or provide a new one...
The user
  • Or provide a new one...
Either the application or the user
  • Or provide a new one...
A MediaStreamTrack can be disabled by the application.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
'readonly' and 'remote' are attributes on a MediaStreamTrack.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
Opus is enabled by default on all WebRTC browsers.
  • Or provide a new one...
FALSE
  • Or provide a new one...
TRUE
  • Or provide a new one...
In general, no STUN servers need to be provided for WebRTC applications to be interoperable because all browsers have default STUN servers configured.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
There are currently two different implemented ways for sending a stream to an HTML media element.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
All WebRTC implementations today implement the standards as written.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
For WebRTC applications, interoperability is the ability for an application running on one device/platform to interact with another.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
For WebRTC applications, portability is the ability of an application designed for one platform to run on another.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
With WebRTC, it's possible to build a peer-to-peer application to connect two people merely by having them load the same URI in their browser.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
The browser will automatically figure out which side should place the offer and which the answer.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
Once the signaling channel is set up, the audio and video are live and being transmitted.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
An XMLHTTPRequest object can receive _______ data.
  • Or provide a new one...
only XML
  • Or provide a new one...
only binary
  • Or provide a new one...
only JSON strings
  • Or provide a new one...
arbitrary
  • Or provide a new one...
When using XMLHTTPRequest for signaling, all requests for a complete WebRTC session usually travel over a single XHR session.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
To reduce feedback, local microphone playback may need to be muted.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
Typically, a key function of the signaling channel is to locate the peer (the other browser).
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
Web server-based signaling channels are usually ______
  • Or provide a new one...
polling-oriented
  • Or provide a new one...
either polling-oriented or session-oriented
  • Or provide a new one...
session-oriented
  • Or provide a new one...
neither polling- nor session-oriented
  • Or provide a new one...
How do two WebRTC browsers initially find each other?
  • Or provide a new one...
application-dependent but signaling server can help
  • Or provide a new one...
the browsers take care of it automatically
  • Or provide a new one...
only via WebSockets
  • Or provide a new one...
The web server providing the HTML and JavaScript can also act as the signaling server.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
setRemoteDescription() must be called ______ calling createAnswer().
  • Or provide a new one...
before
  • Or provide a new one...
after
  • Or provide a new one...
as part of
  • Or provide a new one...
Your code must receive a media description (either offer or answer) from the remote browser and call setRemoteDescription() in order for a call to occur.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
Your code must send a media description (either offer or answer) to the remote browser for a call to occur.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
To initiate a call, your code will likely call createOffer() and then setLocalDescription().
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
srcObject is a new property on audio and video elements to which a MediaStream can be assigned.
  • Or provide a new one...
FALSE
  • Or provide a new one...
TRUE
  • Or provide a new one...
The srcObject object property is used to convert a MediaStream into a URL for printing out to the user.
  • Or provide a new one...
FALSE
  • Or provide a new one...
TRUE
  • Or provide a new one...
When is the onaddstream handler executed?
  • Or provide a new one...
When the local browser has added a stream
  • Or provide a new one...
Both of the above
  • Or provide a new one...
Neither of the above
  • Or provide a new one...
When the remote browser has added a stream
  • Or provide a new one...
When is the onicecandidate handler executed?
  • Or provide a new one...
When the local browser identifies another candidate address at which it can be reached
  • Or provide a new one...
When the remote browser identifies another candidate at which it can be reached
  • Or provide a new one...
When your code calls addIceCandidate()
  • Or provide a new one...
The Peer Connection must be set up before obtaining local media.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
Local media must be obtained before setting up the Peer Connection.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
Media begins flowing after _______.
  • Or provide a new one...
setLocalDescription() and setRemoteDescription() have been called
  • Or provide a new one...
createOffer() and createAnswer() have been called
  • Or provide a new one...
createAnswer() has been called
  • Or provide a new one...
createOffer() has been called
  • Or provide a new one...
all of these
  • Or provide a new one...
The browser automatically handles all media negotiation with the other browser, with no need for application code.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
SDP offer/answer is the approach used to exchange media descriptions between two browsers.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
Media must be attached to a Peer Connection using addStream() in order for it to be sent over the connection.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
addStream() is how you start sending media across a Peer Connection
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
An RTCPeerConnection provides support for ________.
  • Or provide a new one...
Setting up a connection between two browsers
  • Or provide a new one...
Sending media between two browsers
  • Or provide a new one...
Sending data between two browsers
  • Or provide a new one...
All of the above
  • Or provide a new one...
A Peer Connection is established automatically by the browser as soon as local media is obtained.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
The getUserMedia() call returns a MediaStream object containing one or more MediaStreamTrack objects.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
The getUserMedia() call ______.
  • Or provide a new one...
Starts media flowing
  • Or provide a new one...
Sets up a Peer Connection
  • Or provide a new one...
Neither of these
  • Or provide a new one...
All track statistics returned via the Statistics API are 'point in time' values that can be compared over time to determine rate amounts.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
getIdentityAssertion() can be used to start the process of identity verification
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
Activation of the Identity verification mechanism in WebRTC only happens when the application developer uses the APIs.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
The Identity API allows a third party authentication system such as Facebook Connect or OAuth to be used to verify the end user.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
What is the point of a data channel that is not reliable and does not provide messages in order?
  • Or provide a new one...
It's simple
  • Or provide a new one...
It doesn't lose messages
  • Or provide a new one...
It's fast
  • Or provide a new one...
All of the above
  • Or provide a new one...
A user permission check is necessary to create a data channel.
  • Or provide a new one...
TRUE
  • Or provide a new one...
FALSE
  • Or provide a new one...
A data channel can provide ________ message delivery.
  • Or provide a new one...
Reliable and ordered
  • Or provide a new one...
Reliable and unordered
  • Or provide a new one...
Unreliable and ordered
  • Or provide a new one...
Unreliable and unordered
  • Or provide a new one...
All of the above
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All data channels provide reliable, ordered message delivery.
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TRUE
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FALSE
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Each data channel in an RTCPeerConnection is sent as a separate data flow at the protocol level.
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TRUE
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FALSE
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DTMF tones are sent via methods on an AudioStreamTrack.
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TRUE
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FALSE
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DTMF can be added to any audio stream sent over a Peer Connection.
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TRUE
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FALSE
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RFC 4733 DTMF information is sent over (S)RTP.
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TRUE
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FALSE
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The DTMF, Data, Statistics, and Identity APIs are all part of the RTCPeerConnection API.
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TRUE
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FALSE
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addStream() is the method that starts media being sent across a Peer Connection.
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FALSE
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TRUE
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addStream() is used to tell the Peer Connection about a stream to be sent.
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TRUE
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FALSE
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Which methods generate SDP media descriptions?
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setLocalDescription() and setRemoteDescription()
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createOffer() and createAnswer()
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All of the above
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addIceCandidate() is used to tell the browser about candidate addresses for the remote browser.
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TRUE
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FALSE
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No configuration options are available to control which kinds of ICE servers are used.
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TRUE
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FALSE
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The list of ICE servers can be specified when creating an RTCPeerConnection.
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TRUE
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FALSE
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SDP stands for the Session Description Protocol.
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TRUE
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FALSE
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If the browser has received an offer but not yet generated an answer and it receives another offer, this will cause an error.
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TRUE
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FALSE
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For media to begin flowing, the browser needs to have ________.
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A local description
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A remote description
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Both a local description and a remote description
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Neither a local nor a remote description
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Calling setLocalDescription() to give the browser the offer or answer we generated locally is all that is necessary for media to begin flowing.
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TRUE
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FALSE
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The offer contains _______.
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ICE Candidates
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Description of media and data to send
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Acceptable codecs to use
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All of the above
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Media negotiation in WebRTC between two browsers uses an offer/answer approach.
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TRUE
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FALSE
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The difference between disconnected and failed in the ICE connection state machine is that failed is temporary but disconnected is permanent.
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TRUE
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FALSE
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The difference in ICE processing between connected and completed is that with the former new paths could still be found while with the latter state no more paths are available to be checked.
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TRUE
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FALSE
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An example successful path through the ICE connection state machine is __________ .
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New;checking;connected;completed
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New;checking;closed
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New;checking;failed
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Gathering begins when the offer or answer is requested by the JavaScript code.
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FALSE
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TRUE
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Gathering and checking are the same thing in ICE processing
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TRUE
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FALSE
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A Peer Connection is a container object for _________.
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An ICE state machine
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An offer/answer state machine
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A description of local and remote media to be transmitted
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All of the above
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User permissions are only persistent when HTTPS is used for the page.
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TRUE
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FALSE
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A valid constraint in getUserMedia() is ______.
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{mandatory:{audio:true}}
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{mandatory:{video:true}}
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{mandatory:{audio:true, video:true}}
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All of the above
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getUserMedia() is an asynchronous method call.
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TRUE
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FALSE
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It is possible to specify handlers for when tracks are added to or removed from a MediaStream.
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TRUE
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FALSE
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The same exact track can be added twice to a MediaStream
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TRUE
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FALSE
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A MediaStream can only be obtained from getUserMedia()
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TRUE
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FALSE
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All tracks within a MediaStream are required to be synchronized.
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TRUE
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FALSE
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A MediaStream can only contain audio or video, but not both
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TRUE
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FALSE
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getUserMedia() returns a _________.
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MediaStreamTrack
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RTCPeerConnection
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MediaStream
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MediaStreamChannel
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All of the above
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The purpose of getSourceInfos on a MediaStreamTrack is to allow the application to see the list of named devices available for media capture.
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FALSE
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TRUE
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No error or event is generated if the browser cannot satisfy an optional constraint.
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TRUE
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FALSE
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No error or event is generated if the browser cannot satisfy a mandatory constraint.
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TRUE
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FALSE
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The constraint approach was developed for use with media selection and configuration but can now be used to specify other things such as network properties for Peer Connections.
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FALSE
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TRUE
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With constraints, application developers can ________
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Specify optional prioritized constraints
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Specify mandatory constraints
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Specify at the level of precision that matters to them
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All of the above
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Constraints provide a compromise between 'browser knows best' and 'developer knows best'
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TRUE
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FALSE
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Peer-to-peer media flows established between browsers is complicated due to the presence of ______
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NATs
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DTLS
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TURN Servers
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STUN Servers
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If SIP is used as the signaling protocol for WebRTC, a SIP Proxy Server must be used.
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TRUE
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FALSE
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______ provide a way to set video resolution, size, etc. while allowing the browser flexibility in how it does it.
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Objects
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Offers
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Constraints
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Methods
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The current value of a source's constrainable property is called its ______
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Condition
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Form
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Setting
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Position
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_________ describe the possible values for constraints
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Methods
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States
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Capabilities
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Objects
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________ is the method to use to get access to a local camera or microphone.
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setDevice()
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addStream()
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connect()
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getUserMedia()
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getUserMedia() requires verification of user ________.
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Devices
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Consent
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Status
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Connectivity
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In ICE processing, the gathering process refers to the collection of _________ addresses.
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STUN
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TURN
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NAT
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Candidate
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None of the above
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The reply to the offer is called the _______ and contains everything the offer does, plus the streams to be received
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Constraint
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Answer
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Candidate
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Response
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Offers and answers are exchanged via the _________ channel
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Data
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Peer
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Signaling
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To deal with a new remote stream being added, the JavaScript code would set the __________ handler.
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onaddstream
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onnegotiationneeded
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oniceconnectionstatechange
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onsignalingstatechange
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The maxRetransmitTime and maxRetransmits attributes on RTCDataChannel are used to control the _______ of message delivery.
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Speed
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Reliability
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Security
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Privacy
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The _______ API provides a way to get information on the number of bytes and packets transmitted and received on a track over a Peer Connection.
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Data
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Indicators
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Info
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Statistics
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Capacities
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Local media is obtained using the ________ method call.
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getLocalStreams
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addStream
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getUserMedia
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getStreamById
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setLocalDescription
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getAudioTracks() and getVideoTracks() are used to get the tracks from a ______.
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Stream
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Channel
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Path
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Constraint
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In WebRTC applications, the first communication between two web browsers is usually via the signaling channel
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TRUE
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FALSE
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Because they can be time consuming, it may be a good idea to obtain local media and set up the signaling channel __________, in parallel.
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serially
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uncontemporaneously
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allochronically
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asynchronously
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