I'm looking to feed Output from Hardware encoder (H264) to twilio webrtc. Is this possible at all ? Any sample code is helpful.
I've run similar sample code from AWS kinesis video streams. It lets you feed H264 encoded frames: https://github.com/awslabs/amazon-kinesis-video-streams-webrtc-sdk-c/tree/master/samples
Related
I use getUserMedia to the the audio stream, and pass the stream into web audio using createMediaStreamSource. I then want to stream the raw audio data into a webrtc data channel.
There isn’t a data channel destination node in web audio. I’ve only been able to access the raw audio data from inside an audio worklet, but I don’t know how to get that data into a data channel. How should I go about streaming raw audio from getUserMedia data into a data channel?
Can someone provide training or documentation about how actually video flows between two sip clients. I know basics of where client apis but don’t have much knowledge on how rtp packets are formatted , send over wire and received by other client
What are the header and what happens during packet loss
And how does a video gets converted into a rtp packet
Colin Perkins book "RTP: Audio and Video for the Internet" book is, despite being published in 2003, still the best guide for this.
Suppose we get an audio stream by using getUserMedia(gUM) API. I am broadcasting this stream to other users using WebRTC.
Is it possible to reduce the volume of the audio getting streamed.
Note : I am not looking to reduce device mic volume, because I understand we cannot control through browser and I dont want to
Try with GainNode from WebAudio. But the best thing will be to do this on the receiving end.
I'm new in NAudio and Audio Programming, and I have questions about streaming in 256kbps, NAudio have some codec to work with this bitrate? I know about MP3 Encode in NAudio don't support. Can I use other codec like ACC in NAudio or another option with this quality to encode?
I'll need to streaming in first side and receive streaming in application on the other side (unicast).
Some ideas?
Thanks!
we are using zte usb modem. we try to call by AT command (ATD) successfully. But there is no sound when remote device answered.
Does anyone have any idea?
My problem was associated with ZTE usb modem.
I solved the problem.
i can receive and send voice separately to voice port now. But i can not get clean sound like WCDMA UI.
how can i receive and send data with high quality?
Please look at my source code. [http://serv7.boxca.com/files/0/z9g2d59a8rtw6n/ModemDial.zip]
Does anyone now where is my error?
Thank you for your time.
a) Not all zte usb modems supports voice, to detect if modem supports check for ZTE voUSB Device in your ports list.
b) If port present, voice will go through it in pcm format, with 64kbps frequency (8000 samples per sec, 8 sample size).
In your own program, you should read audio stream from there.
stream is additionaly encoded with g.711, so you need to decode it before sending to audio device
It is fairly common to shut off the speaker after connection. Try sending ATM2, that should make the speaker always on.
Basic hayes command set:
M2
Speaker always on (data sounds are heard after CONNECT)
I'm trying to use asterisk's chan_dongle module on ZTE MF180 Datacard model with activated voices abilities.
Originally chan_dongle using raw PCM format on voice data.
But i was discover, that ZTE using ulaw format on sending and recciving voice data.
You can get voice data and save file in this format for learn by using standard Asterisk's Record(filename:ulaw) command in dialplan.
My voice data, dumped from ZTE modem in the same format.
I check it. ZTE dumped data was successefully played by Asterisk's command Playback(dumped)