Re: RTP over websockets

[Vikas]

Hi,

AFAIK, currently you can use websockets for webrtc signaling but not
for sending mediastream.

/Vikas

On Apr 25, 3:29 am, jirikut...@googlemail.com wrote:
> Folks,
>
> sorry for a beginner question but is there a way for webrtc apps to send
> RTP/SRTP over websockets?
> (as the last-resort method for firewall traversal)?
>
> thanks!
>
> jiri

[Bryan]

You could peer with a server using TCP candidates and have something that looks almost the same as websockets.  

Unfortunately, TCP candidates ignore the configured proxy server and take 2 seconds to connect.

I'm really hoping that these two issues will get resolved soon.

[jirik...@googlemail.com]

Still would not it make sense to allow for overloading the "normal" sockets by websockets?

I mean the concern I have (and couple of folks I have incidentally talked to as well) is, that 

without possibility to do web tunneling one cannot really match skype. That's something

which would be kind of bizzar after all the years of shorttalks, when I tell "I do VoIP" and the

guys I talk to answer "I do skype too" :) At least I think this SIP failure has been

largely  consequence of ignoring NATs/firewalls.

OTOH -- some don't like web sockets as policy circumvention vehicle. That's why I have been 

hopeful there would be some possibility like overloading regular sockets with websockets.

This would really leave the option whether to tunnel or not to the JAvascript programmer,

who either wishes to get over everything  (including my DSL home routers that have some

concealed broken ALGs) and use HTTP tunnels as last resort, or comply to IT policies.

-jiri

[Kaiduan Xie]

With the support of UDP port allocation over TCP connection (browser establishes a TCP connection to TURN server, and requests TURN server to allocate UDP port for it), webRTC will traverse firewall where only certain outbound TCP ports are allowed in enterprise network.

/Kaiduan

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[medusade]

RTP is intended to be used over fast connectionless networking via UDP, where as websockets uses TCP connections for socket virtualization. If implemented the performance for video traffic would probably be poor over websockets.

medusade 

[Kaiduan Xie]

We have tested the audio/video over TCP connection to TURN server, the quality is decent.

/Kaiduan

--

[Oleg Moskalenko]

You are right, a TURN server with TCP support allows to work it out without websockets.

But "pure" UDP (through TURN server), if available, will be better than TCP, in terms of quality.

Oleg

[Lorenzo Miniero]

I proposed some kind of HTTP tunneling fallback for RTP in a (now expired) draft a few months ago, but it didn't raise much interest in RTCWEB. Besides, a few people pointed out that solutions that explicitly try and override restrictions in networks are not always welcome, so TURN over 443  is probably as close as you can get to match Skype as a way to traverse such networks.