So there is this Application RetroShare which currently uses speex codec and libav codec for VOIP communications.
They expect a new improved VoIP cross platform codec as the old one aren’t fitting the bill.
What I’ve landed on is implementing VP9 and OPUs codec for better audio and video quality over communications.
How can I approach that ?
Can anyone please tell me is there any tutorial for implementing VP9 codec ? I couldn’t find one!
Or can anyone here tell me how to implement codecs in a Peer-to-peer applications ?
And is my approach right ?
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PJSIP is used for SIP Registering, audio/video Calling and supporting some other VoIP Call features. If you want to create an VoIP Application, you will create an application using their libraries.
In this library, there are available lot of codecs and some of codecs are available as third party codecs and you can integrate into PJSIP library. Then you can able to support those codecs while calling another SIP user/client.
Generally In mobile phones, the voice is record as audio file through
mic and it has passed to PJSIP library. Then those codec/pjmedia library take care
of other operations like converting into anyother codec format which
they want to send.
Instead of that, can we able to pass .mp3/.wav file into PJSIP library and convert it into codec files like .g729/.amr/.g711-u and vice versa
I don't about how codec conversion/codec engine internally works around.
If you know about the codec conversion from .mp3 format is possible using PJSIP library, please suggest me how to solve this problem.
Thanks in Advance!
I been reading a lot about H265 encoder but I'm no really sure how to start a C or Python application to encode a video stream in real time using H.265 encoder from libde265, I all ready install the library and I guess I could use opencv to get the input video stream from a usb camera, do anyone has worked in this type of application ?
If you are not particular about using libde265, which I am not aware of, please give a shot with gstreamer. gstreamer has lots of plugins and examples if you are doing some standard tasks like encoding on your stream. It also integrates well with native development.
I have worked on a project similar to yours, where we did H264 encoding and decoding along with few other things on a live camera feed.
Please find the video of a similar application here: https://www.youtube.com/watch?v=JcpkGDpfVU0
Just my two cents!!
As I understand it, streaming via bluetooth is handled via the A2DP profile. While the SBC codec is default, A2DP supports AAC, MP3, and a few other Codecs.
My question is, since spotify files are in the OGG VORBIS format (OGG Container, Vorbis Codec), what is the best way to handle streaming via Bluetooth without quality loss? Is there a specific A2DP implementation? Are folks like Jambox, etc just using the SBC implementation?
Spotify's streaming format is an implementation detail to all clients, and making the assumption that it's OGG Vorbis is not something you should do, and in some circumstances is actually a false assumption.
Since you've managed to use every single Spotify tag in your question, I don't know which platform you're developing for. However, the correct thing to do is take the PCM data the Spotify playback library gives you and use whatever playback stack your target platform gives you. On Android, iOS, Mac OS, etc the system will handle audio output devices for you, including Bluetooth streaming.
i have just started to delve into streaming libraries and the underlying protocols. I understand rtsp/rtp streaming and what these 2 protocols are for. But if we need the ip address, codec and the rtsp/rtp protocols to stream the video and audio from any cameras then why do we have onvif standard which essentially also aims to standardize the communication between IP network devices. I have seen the definitions of onvif so thats not what I am looking for. I want to know why at all we need onvif when we already have rtsp/rtp and what additional benefits it can provide.
ONVIF is much more than just video streaming. It's an attempt to standardize all remote protocols for network communication between security devices. This includes things like PTZ control video analytics and is much more than just digital camera devices.
Is it possible to play custom audio (*.wav file) over VOIP (SIP) without sound card being installed on SIP client machine? All my needs is to perform SIP call and play custom sound message.
You can transmit a recorded audio in the form of a WAV file over a SIP signal if you convert it to the appropriate codec first. This does not require a sound card to transmit this audio. A sound card is only required to listen to the audio. Which codec to use depends on the platform. Here is a link for converting to appropriate codecs when using Asterisk. There are a lot more if you just Google something like "audio codec conversion".
A simpler approach is to just use a platform that does this for you, like Voxeo Prophecy. This is a software only IVR solution that has a 2 port version for free. It is easy to install and program using the open standard VoiceXML. It will play back audio files recorded in a WAV file format and the telephony interface is SIP.