webrtc-sdk | WebRTC Simple Calling API + Mobile SDK | SDK library

 by   stephenlb JavaScript Version: 2.2.7 License: MIT

kandi X-RAY | webrtc-sdk Summary

kandi X-RAY | webrtc-sdk Summary

webrtc-sdk is a JavaScript library typically used in Utilities, SDK applications. webrtc-sdk has no bugs, it has no vulnerabilities, it has a Permissive License and it has medium support. You can install using 'npm i webrtc-sdk' or download it from GitHub, npm.

WebRTC SDK Upgraded! ES6, new camera control and 100x less code than v1. The following demo uses PubNub for signaling to transfer the metadata and establish the peer-to-peer connection. Once the connection is established, the video and voice runs on public Google STUN/TURN servers. Keep in mind, PubNub can provide the signaling for WebRTC, and requires you to combine it with a hosted WebRTC solution. For more detail on what PubNub does, and what PubNub doesn’t do with WebRTC, check out this article: At PubNub we believe simplicity is essential for our SDK usability. We've taken a simplified approach to WebRTC Peer Connections by creating and easy-to-use SDK for developers. The ideas of simplicity should span all platforms and devices too and that's why we also support Android WebRTC mobile calling with compatibility for iOS native Objective-C based WebRTC SDK. This simple developer WebRTC SDK is powered by PubNub Data Stream Network.
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            kandi-support Support

              webrtc-sdk has a medium active ecosystem.
              It has 832 star(s) with 281 fork(s). There are 55 watchers for this library.
              OutlinedDot
              It had no major release in the last 12 months.
              There are 28 open issues and 37 have been closed. On average issues are closed in 135 days. There are no pull requests.
              It has a neutral sentiment in the developer community.
              The latest version of webrtc-sdk is 2.2.7

            kandi-Quality Quality

              webrtc-sdk has 0 bugs and 0 code smells.

            kandi-Security Security

              webrtc-sdk has no vulnerabilities reported, and its dependent libraries have no vulnerabilities reported.
              webrtc-sdk code analysis shows 0 unresolved vulnerabilities.
              There are 0 security hotspots that need review.

            kandi-License License

              webrtc-sdk is licensed under the MIT License. This license is Permissive.
              Permissive licenses have the least restrictions, and you can use them in most projects.

            kandi-Reuse Reuse

              webrtc-sdk releases are not available. You will need to build from source code and install.
              Deployable package is available in npm.
              Installation instructions are not available. Examples and code snippets are available.
              It has 863 lines of code, 0 functions and 15 files.
              It has low code complexity. Code complexity directly impacts maintainability of the code.

            Top functions reviewed by kandi - BETA

            kandi has reviewed webrtc-sdk and discovered the below as its top functions. This is intended to give you an instant insight into webrtc-sdk implemented functionality, and help decide if they suit your requirements.
            • Create socket . io socket
            • creates a conversation
            • Main Requester .
            • connect to another channel
            • receive message from server
            • Adds a offer to offer .
            • setup snapshot of snapshot
            • Read the stream
            • Publish the send queue .
            • Recursively merge objects .
            Get all kandi verified functions for this library.

            webrtc-sdk Key Features

            No Key Features are available at this moment for webrtc-sdk.

            webrtc-sdk Examples and Code Snippets

            No Code Snippets are available at this moment for webrtc-sdk.

            Community Discussions

            QUESTION

            cmake find_library() not finding library specified by PATHS
            Asked 2021-Apr-22 at 23:12

            I have a find_library() statement that matches below (this is based on the amazon kinesis project):

            find_library(SRTP_LIBRARIES NAMES srtp2 REQUIRED PATHS ${OPEN_SRC_INSTALL_LIB_PREFIX})

            The OPEN_SRC_INSTALL_LIB_PREFIX correctly points to the location where this library is located. I can observe this directly. However, this find_library() call fails and I am confused as to why it would do so.

            I thought that maybe cmake is searching through other paths first and ignoring my specified path, so I also tried it with the NO_DEFAULT_PATH flag, as that should limit the search to only be in the paths specified:

            find_library(SRTP_LIBRARIES NAMES srtp2 REQUIRED PATHS ${OPEN_SRC_INSTALL_LIB_PREFIX} NO_DEFAULT_PATH)

            ... still no luck.

            Any ideas why this would not work? I've verified the OPEN_SRC_INSTALL_LIB_PREFIX is the valid directory of the library via message() prints.

            Note that this is cross compiling, although I don't see why that would change the behavior of find_library() unless I'm missing something from the documentation

            ...

            ANSWER

            Answered 2021-Apr-22 at 23:12

            First want to note I'm on cmake 17.5.

            Ok, so the documentation is a bit confusing, but it seems that the NO_DEFAULT_PATH would only use the paths specified by the PATHS argument and none of the other cache variables, but that doesn't seem to be the case if CMAKE_SYSROOT is set.

            Using the NO_CMAKE_FIND_ROOT_PATH is what actually causes find_library() to ignore the cached paths.

            Source https://stackoverflow.com/questions/67130750

            QUESTION

            library link error when cross-compiling aws webrtc library
            Asked 2021-Jan-20 at 08:42

            I'm trying to cross compile aws webrtc library for arm but I encountered an error while linking C excutable test_srtp. I mixed the instruction of Readme.md and some stackoverflow post for the command. But I am still stuck here for 1 week and I can not find the case on google. How can I fix it?

            What command I input

            ...

            ANSWER

            Answered 2021-Jan-20 at 08:42

            I just solved it by using mbedtls instead of OpenSSL. Try it with mbedtls if there is someone who have the same issue.

            Source https://stackoverflow.com/questions/65729924

            QUESTION

            failed cross compiling aws webrtc library
            Asked 2021-Jan-11 at 07:53

            I'm trying to cross compile aws webrtc library(https://github.com/awslabs/amazon-kinesis-video-streams-webrtc-sdk-c) for arm. So I followed the instruction on Readme.md but I failed.

            What I did

            ...

            ANSWER

            Answered 2021-Jan-07 at 10:25

            Something is wrong with your ARM toolchain or how the build system uses it. Where did you get it from and how did you install it?

            The header sys/capability.h is included with the ARM toolchain. For example, when I install the Debian package gcc-arm-linux-gnueabihf then this header file gets copied to /usr/arm-linux-gnueabihf/include/linux/capability.h. The toolchain is supposed to find it there on its own.

            The command arm-linux-gnueabihf-gcc -v will show you where the toolchain is configured to look for its own header files. E.g. in output below it's --includedir=/usr/arm-linux-gnueabihf/include that matters, the header should be found relative to this directory. I apologize for the super long line.

            Source https://stackoverflow.com/questions/65608339

            QUESTION

            C Struct in Swift
            Asked 2020-Sep-28 at 16:43

            I'm attempting to use amazon-kinesis-video-streams-webrtc-sdk-c in a Swift project however I'm unsure of how to create a SignalingClientInfo struct. Specifically I'm unsure of how to properly construct the clientId:

            ...

            ANSWER

            Answered 2020-Sep-28 at 16:43

            Kind of annoying, but fixed-sized arrays in C like clientId are imported into Swift as tuples.

            There's on-going discussions on adding proper fixed-size arrays to Swift, but in the mean time, there are implementation-dependent tricks you can use to construct large tuples from arrays.

            See https://oleb.net/blog/2017/12/swift-imports-fixed-size-c-arrays-as-tuples/

            Edit: it looks like the layout of homogeneous tuples is guaranteed. So this is safe, just annoying.

            Source https://stackoverflow.com/questions/63603473

            QUESTION

            Handle network interruption in WebRTC
            Asked 2020-Sep-19 at 09:16

            I am using Amazon Kinesis Video Streams WebRTC for video chat connection between a mobile app and web site. I need to handle network interruption on web side. Let me explain the steps I have done already.

            I can detect that network is disconnected on event listener ...

            ...

            ANSWER

            Answered 2020-Sep-19 at 09:16

            I found that I had to send sdp offer. So the code will be like ...

            Source https://stackoverflow.com/questions/63848245

            QUESTION

            WebRTC signalling succeeds but no video is coming through
            Asked 2020-Aug-31 at 12:40

            I am trying to setup a video stream from a robot using WebRTC.

            As far as I can tell, the signaling establishes a connection successfully, however no video is being streamed.

            I use AWS Kinesis Video as the signaling server, and I use the AWS Kinesis video streams WebRTC sdk for the master node.

            For the viewer, I use the Kinesis WebRTC Test Page, with the only change that the viewer does not request any audio.

            I believe that the signaling works. At least the viewer is both sending and receiving ICE candidates and there are no errors.

            The master node also starts to send data as expected, but the video is never displayed on the viewer.

            My question is how can I debug where the problem is?

            I have looked at Chrome's webrtc internals, and it generated these graphs. This shows me that packages are being received by chrome, but no frames are being decoded. Is that correct?

            On the robot, I am running RoS as the middlelayer, and I am trying to stream the usb webcam.
            To do this, I am running the h264_video_encoder node. When the nodes starts up, it tells me that it has these settings:

            264 - core 152 r2854 e9a5903 - H.264/MPEG-4 AVC codec - Copyleft 2003-2017 - http://www.videolan.org/x264.html - options: cabac=1 ref=1 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=2 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=0 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=3 lookahead_threads=3 sliced_threads=1 slices=3 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=1 keyint=30 keyint_min=16 scenecut=40 intra_refresh=0 rc=abr mbtree=0 bitrate=2048 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00

            I suspect the most likely problem is with the encoding, but I don't know how to proceed with debugging this issue.

            ...

            ANSWER

            Answered 2020-Aug-31 at 12:40

            The problem that I faced was that the h264 stream I generated was missing the SPS and PPS NALs, so the viewer did not know how to decode the stream.

            The underlying problem was that I was converting from KinesisVideoFrame.msg which contained my encoded stream, but I was unaware that the frame_data did not contain the SPS and PPS packets. Those packets are stored in codec_private_data, so I had to prepend codec_private_data to the frame_data to get the final frame data to be send over the stream

            Source https://stackoverflow.com/questions/63617797

            QUESTION

            WebRTC Android echo cancellation
            Asked 2020-Jun-22 at 16:28

            I would like to implement voice and video calling in android. I used this example project as reference: Amazon Kinesis Video Streams Android WebRTC SDK Problem is that the microphone picks up the remote sounds which results in an extreme echoing effect. We use this version:

            ...

            ANSWER

            Answered 2020-Jun-22 at 16:28

            Use below methods of JavaAudioDeviceModule.class

            Source https://stackoverflow.com/questions/62479789

            Community Discussions, Code Snippets contain sources that include Stack Exchange Network

            Vulnerabilities

            No vulnerabilities reported

            Install webrtc-sdk

            You can install using 'npm i webrtc-sdk' or download it from GitHub, npm.

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