pcma | Page Cache My Assets | Caching library

by pcarrier C Version: Current License: ISC

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kandi X-RAY | pcma Summary

pcma is a C library typically used in Server, Caching applications. pcma has no bugs, it has no vulnerabilities, it has a Permissive License and it has low support. You can download it from GitHub.

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Support

pcma has a low active ecosystem.

It has 10 star(s) with 4 fork(s). There are no watchers for this library.

It had no major release in the last 6 months.

pcma has no issues reported. There are no pull requests.

It has a neutral sentiment in the developer community.

The latest version of pcma is current.

Quality

pcma has no bugs reported.

Security

pcma has no vulnerabilities reported, and its dependent libraries have no vulnerabilities reported.

License

pcma is licensed under the ISC License. This license is Permissive.

Permissive licenses have the least restrictions, and you can use them in most projects.

Reuse

pcma releases are not available. You will need to build from source code and install.

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pcma Key Features

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pcma Examples and Code Snippets

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Community Discussions

Trending Discussions on pcma

JSON type NSMutableDictionary cannot be converted to .sdp must not be null

SessionDescription is Null. in web rtc after updating Chrome to latest v 89. Working on previous Chrome Versions

WebRTC browser peerconnection depends on who creates an offer

Why the caller don't hear echo when I'm relpying the RTP packets I receied

Set Max Bandwidth Of Kurento media server over 500kb for webrtc

How to remove noise added when converting pcma/aluw file I received in RTP to wav?

React Native + WebRTC TypeError: Undefined is not and Object(Evaluating: 'data.type')

WebRTC Client side can not display remote stream on video element

Webrtc and UWP without trickle ice

Android WebRTC low success connection rate

QUESTION

JSON type NSMutableDictionary cannot be converted to .sdp must not be null

Asked 2021-Jun-05 at 06:38

Am new to webRTC and am trying to create a react native app with video calling functionality using this tutorial here as an example to follow https://dipanshkhandelwal.medium.com/video-calling-using-firebase-and-webrtc-14cc2d4afceb

However i keep getting this error on iOS and on android the app just closes once i try to join a call. The error i get on iOS says:

...

ANSWER

Answered 2021-Jun-05 at 06:38

I guess you are trying to use firebase as a signalling medium and want to use react-native-webrtc for the video calling.

Here is the sample code I have for the same solution with the latest libraries and react-native version.

Firebase Installation React Native.

Just set up ios and android using this above link and then use the below code for reference.

Source https://stackoverflow.com/questions/67671361

QUESTION

SessionDescription is Null. in web rtc after updating Chrome to latest v 89. Working on previous Chrome Versions

Asked 2021-Mar-22 at 15:14

"Answer" of SDP offer is not sent to Web App(running on Windows/Mac/Linux) from Android/iOS application after updation of Chrome to latest v89 (released on 9 March 2021).

It is working fine on Chrome v88 and below.

Offer SDP:

...

ANSWER

Answered 2021-Mar-22 at 15:14

The issue is caused by Chrome enabling extmap-allow-mixed by default (see https://www.chromestatus.com/features#offerExtmapAllowMixed). It can be solved by the receiver mangling the offer SDP to remove the line containing a=extmap-allow-mixed

Source https://stackoverflow.com/questions/66641204

QUESTION

WebRTC browser peerconnection depends on who creates an offer

Asked 2021-Feb-12 at 14:12

Two Chrome browsers: Alice (A), Bob (B). Different networks, so i'm using Coturn server (my own).

The problem is that when A creates an offer - everything is ok, ice connection state goes to "connected", everything works fine. But if B creates offer - every peer receives the same Ice candidates, but ice connection state after 10 sec "checking" goes to "disconnected". It depends on in what network is B. Only on some networks there is such problem.

Here are the details:

Not working case:

B creates an offer. His descriptor is:

...

ANSWER

Answered 2021-Feb-12 at 14:12

You are not getting any candidates with typ relay which means you are only using your TURN server as a STUN server. There are a couple of NATs where that can lead to a failure to establish the connection depending on who offers. There is an open bug in chrome's webrtc implementation for a couple of years now here

A working TURN server avoids the problem.

Source https://stackoverflow.com/questions/66163961

QUESTION

Why the caller don't hear echo when I'm relpying the RTP packets I receied

Asked 2021-Feb-08 at 12:13

I'm trying to play audio for who is calling me all by using NodeJS.

He is using SIP and initiate a call. I was able to receive the call and record his audio.

Now I'm trying to use the RTP protocol to play back audio to him.

So what I tried is to save each RTP packet I got, and send it back as is. But I'm don't undestand why he don't hear nothing.

The INVITE command

...

ANSWER

Answered 2021-Feb-08 at 12:13

In other words: According to RTP protocol, is sending back the packet can work, or do I need to reconstruct them?

You need to construct a dedicated RTP header and add the received payload. SSRC and timestamp must be generated in a meaningful way or the receiver may discard those packets.

Source https://stackoverflow.com/questions/66055522

QUESTION

Set Max Bandwidth Of Kurento media server over 500kb for webrtc

Asked 2021-Jan-24 at 15:00

Hiii

I am using Kurento media server in my webrtc Project. I am want to Increase Video Quality for my web call.

setmaxsend/recbandwith() is not working for value over 500 kb.

I want to change max bandwidth tell me how can do that.

what I need:

Is there any way to to that
Can i find actual variable which is used to define the bandwidth.
I want to Set Max Bandwidth 2000kb.

My Sdp

...

ANSWER

Answered 2021-Jan-21 at 22:31

I changed the SDP bandwidth attribute when I went to create an answer:

Source https://stackoverflow.com/questions/65806623

QUESTION

How to remove noise added when converting pcma/aluw file I received in RTP to wav?

Asked 2021-Jan-12 at 09:22

This is the sdp:

...

ANSWER

Answered 2021-Jan-12 at 09:20

You should use the -alaw format instead. And it will work for you.

Source https://stackoverflow.com/questions/65680907

QUESTION

React Native + WebRTC TypeError: Undefined is not and Object(Evaluating: 'data.type')

Asked 2020-Jun-02 at 17:37

I've been trying to learn WebRTC with React Native, so I've been working on this video call project. I thought I had it all figured out until this came up.

First of all, this is my React Native Code

...

ANSWER

Answered 2020-Jun-02 at 17:37

There is a typo in onmessage on React Native. rename dcata to data. upvote if you find the answer useful.

Source https://stackoverflow.com/questions/62157057

QUESTION

WebRTC Client side can not display remote stream on video element

Asked 2020-May-04 at 10:35

I am trying to test a WebRTC application for video calling whose signalling server is written in Spring Boot. This is the client code that I am running. However in case of an offer, the local stream on the remote window gets opened and I can see the cam view in the local view video element but the remote view element is just blank. I think i have made some obvious mistakes in my code. Please help me out. Client code :

...

ANSWER

Answered 2020-May-04 at 10:35

Its onaddstream, all lower-case. Also note that onaddstream has been deprecated in favor of ontrack.

Source https://stackoverflow.com/questions/61582809

QUESTION

Webrtc and UWP without trickle ice

Asked 2020-Mar-01 at 10:27

I'm trying to establish a webrtc call using this nuget package https://www.nuget.org/packages/WebRtc/. I am not able to use trickle ice in my implementation so I try to add ice candidates to my sdp as they are gathered. This worked in an earlier version of the package, however when using the latest version this fails when I try to create a session description with a modified sdp. The latest version of the package is based on webrtc version m71 as I understand it. My offer sdp looks like this after adding ice candidates:

...

ANSWER

Answered 2020-Mar-01 at 10:27

network-cost 10|audio|0 in the candidate line looks wrong and probably results in a parse error.

Most likely you concatenated the sdpMid and sdpMLineIndex with the candidate when appending to the SDP.

Note that typically the peerconnections localDescription.sdp will be updated by the library without the need to add candidates yourself.

Source https://stackoverflow.com/questions/60464322

QUESTION

Android WebRTC low success connection rate

Asked 2019-Oct-24 at 23:02

I am developing video chat application with native android WebRTC. I am using public google STUN server and XIRRYS STUN and TURN servers.

My problem is that the successfully connection rate is very low !

The success connectivity rate is as follow:

About 50% connection failures when i am using the WIFI network
About 90% connection failures when i am using mobile internet.

In all cases, the offer, answer and candidates messages are running via the signaling server very well but as i say, at meany cases for connection is failed.

i tried to check the SDPs and Candidate information but i cannot see anything problematic there:

here is one example of failure from the LOGCAT:

...

ANSWER

Answered 2019-Oct-24 at 23:02

After many checks and many many test and connection tries.... After so many logs and displays i was able to solved the problem and to achieve 100% successfully connection rate regardless of the network of the devices !

If you are asking, it was very stupid problems (like always in such cases), i have missing UTF8 encoding at the signaling server side, so some data messages was corrupted !

After fixing it, i am getting 100% successfully connection rate !

tfrysinger, For your question, yes I have spend some time with XirSys supoort over their chat but they was not able to help me.

Anyway, my email address is zionrevi@gmail.com.

Thanks, Zion

Source https://stackoverflow.com/questions/57724710

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