codec_silk | asterisk translation module that wraps Skype | Messaging library

by mordak C Version: Current License: No License

X-Ray Key Features Code Snippets Community Discussions(1)Vulnerabilities Install Support

kandi X-RAY | codec_silk Summary

codec_silk is a C library typically used in Messaging applications. codec_silk has no bugs, it has no vulnerabilities and it has low support. You can download it from GitHub.

An asterisk translation module that wraps Skype's SILK codec library

Support

Quality

Security

License

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Support

codec_silk has a low active ecosystem.

It has 14 star(s) with 16 fork(s). There are 3 watchers for this library.

It had no major release in the last 6 months.

There are 0 open issues and 1 have been closed. On average issues are closed in 115 days. There are no pull requests.

It has a neutral sentiment in the developer community.

The latest version of codec_silk is current.

Quality

codec_silk has no bugs reported.

Security

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

License

codec_silk does not have a standard license declared.

Check the repository for any license declaration and review the terms closely.

Without a license, all rights are reserved, and you cannot use the library in your applications.

Reuse

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

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codec_silk Key Features

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codec_silk Examples and Code Snippets

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Community Discussions

Trending Discussions on codec_silk

How do I create Docker container with asterisk 16

QUESTION

How do I create Docker container with asterisk 16

Asked 2019-Jun-13 at 16:00

I want to create a Docker with asterisk 16, I can register my sip extension but I don't listen anything.

I can go to 8088 port too, modifying http.conf Also I edited rtp.conf with

...

ANSWER

Answered 2019-Jun-13 at 16:00

Unfortanly asterisk need AT LEAST 2 ports for ONE call.

For two legs calls(call to provider and back) it already need FOUR ports.

So you need change your rtpsettings and change docker accordinly.

Actually it is REALY bad idea run sip inside docker if you have zero knowledge about SIP. Use some manual from people who have experience.

Source https://stackoverflow.com/questions/56570773

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

Vulnerabilities

No vulnerabilities reported

Install codec_silk

You can download it from GitHub.

Support

Suggestions / bug reports / pull requests are welcome. This is my first time writing anything for asterisk and first repo on github, so please be kind.

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