rfc5766-turn-server | SVN copy of .. ] High-performance free open source TURN | TCP library

by tony2001 C Version: Current License: Non-SPDX

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kandi X-RAY | rfc5766-turn-server Summary

rfc5766-turn-server is a C library typically used in Networking, TCP applications. rfc5766-turn-server has no bugs, it has no vulnerabilities and it has low support. However rfc5766-turn-server has a Non-SPDX License. You can download it from GitHub.

[SVN copy of..] High-performance free open source TURN and STUN Server implementation. VoIP media traffic NAT traversal and gateway.

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rfc5766-turn-server has a low active ecosystem.

It has 19 star(s) with 32 fork(s). There are 6 watchers for this library.

It had no major release in the last 6 months.

There are 1 open issues and 1 have been closed. On average issues are closed in 771 days. There are no pull requests.

It has a neutral sentiment in the developer community.

The latest version of rfc5766-turn-server is current.

Quality

rfc5766-turn-server has no bugs reported.

Security

rfc5766-turn-server has no vulnerabilities reported, and its dependent libraries have no vulnerabilities reported.

License

rfc5766-turn-server has a Non-SPDX License.

Non-SPDX licenses can be open source with a non SPDX compliant license, or non open source licenses, and you need to review them closely before use.

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rfc5766-turn-server releases are not available. You will need to build from source code and install.

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Community Discussions

Trending Discussions on rfc5766-turn-server

Should a turn request include candidates for stun?

WebRTC: Is there a SFU that operates at the transport layer?

QUESTION

Should a turn request include candidates for stun?

Asked 2019-Feb-06 at 08:53

There are two very similar questions here, but both haven't been answered:

rfc5766-turn-server as TURN and STUN for webrtc application question 1. says "with rfc5766-turn-server is enough to act as a TURN and also STUN server or do I also need to run a stun server".

missing relay candidates from TURN server speak about missing relay candidates, the effect I am seeing when making only a turn request.

We have coturn configured with turns and use-auth-secret and use https://webrtc.github.io/samples/src/content/peerconnection/trickle-ice/ for testing.

When I make a stun request stun:hostname:5349 without password I see the srflx candidate (my external IP).
When I make a turn request turns:hostname:5349[username:password] I see the relay candidate (the TURN server's public address).

This looks good, but do I need to make 2 requests to the same server? Shouldn't turn include stun and give me srflx and relay candidates?

Can I configure coturn to return both in a single request?
Or will the client have to make both requests? Is so, will all browsers do this correctly when using WebRTC?

...

ANSWER

Answered 2019-Feb-06 at 08:53

If you have TURN/UDP configured in the client, configuring the same server as a STUN server isn't necessary.

TURN over UDP will return a srflx candidate automatically, you should be able to see that on the trickle-ice test page. If you look at the packets exchanged in Wireshark you'll see both * a STUN binding request and the binding response containing the public ip as seen by the server in the xor-mapped-address field as well as * a STUN (TURN) allocate request whose response also contains a xor-mapped-address

TURN over TLS or TURN over TCP won't do that since that would not work with udp hole punching.

Source https://stackoverflow.com/questions/54546022

QUESTION

WebRTC: Is there a SFU that operates at the transport layer?

Asked 2017-Feb-27 at 22:26

I'm running a WebRTC based service and currently investigating the requirements for WebRTC conference chats with approx. 10 users. Therefore I'd like to run a Selective Forwarding Unit (SFU). I know about janus and jitsi videobridge, but am a little bit concerned about data security. Both servers operate like WebRTC endpoints, which means to me that the incoming streams are de- and encoded before they are passed to the users. Even if it happens only in memory and in a split second, it means a lack of confidentiality (which I have to take serious for legal reasons).

This leads me to my question. Are there any SFU (or multicast relay servers that could be used as SFU) that operate at the transport layer (as described in RFC 7201 chapter 2.3.1, citing RFC 5117) and only pass the SRTP encrypted streams to their recipients?

I thought about (re-) using my TURN server for this purpose, but as Oleg pointed out here the TURN specs are designed for P2P communication only. Maybe I'm just wrong with my assumptions about janus and jitsi videobridge, then I would be glad to here that as well.

Thx in advance

Greg

...

ANSWER

Answered 2017-Feb-27 at 22:26

That's not currently possible, I'm afraid. Advanced SFUs may need to check metadata in the RTP stream, such as to identify a video keyframe, hence the need to access the RTP stream.

But there is hope! The IETF PERC working group has published a draft which allows for end-to-end encryption by applyuing SRTP twice, see it here.

Jitsi is working towards having the first PERC implementation: https://www.slideshare.net/alexpiwi5/perc-webrtc-e2e-media-encryption-with-sfu

Source https://stackoverflow.com/questions/42453958

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