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Which element in webrtc API stat refer to incoming bit rate


Which element in webrtc API stat refer to incoming bit rate

By : silentcloud
Date : November 18 2020, 01:01 AM
help you fix your problem In webrtc-internals check the active connection -- it's printed in bold. Usually it is Conn-Audio-1-0. There are two fields bytesSent and bytesReceived which will allow you to calculate the bitrate. Also check the constraints + stats demo for an actual example: https://webrtc.github.io/samples/src/content/peerconnection/constraints/
In getStats, iterate the reports until you find one of kind googCandidatePair with .stat('googActiveConnection') === 'true'. That is giving you the same information as webrtc-internals. If you want per-track/stream values, reports of type ssrc have bytesSent or bytesReceived, depending on whether they are sent or received.
code :


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Incoming calls with SIP and WebRTC

Incoming calls with SIP and WebRTC


By : POCCoder
Date : March 29 2020, 07:55 AM
will help you You need a server that implements a SIP-WebRTC gateway. The gateway will be able to receive incoming calls from a SIP provider (which itself will be acting as a SIP-PSTN gateway by converting ISDN-SIP, SS7-SIP etc) via SIP and then forward the call to your browser based clients using WebRTC.
Put another way your server needs to be a combination of a SIP server and a HTTP server and the HTTP server needs to support web sockets and the WebRTC API.
View incoming WebRTC connection

View incoming WebRTC connection


By : Tim
Date : March 29 2020, 07:55 AM
With these it helps If you want to actually play the media, there is not a way. It is encrypted with a key that is exchanged in the DTLS handshake at the beginning of the peerconnection.
You can see the UDP packets in wireshark(their source and destination ports) but the media type and actually being able to play it is not possible unless you are privy to the master key exchanged so that you can decrypt the media(which is not possible if you are using the browser javascript APIs).
In Python's stat module, when I print stat.S_IRWXU | stat.S_IRWXG | stat.S_IRWXO, why do I get 511 instead of 777?

In Python's stat module, when I print stat.S_IRWXU | stat.S_IRWXG | stat.S_IRWXO, why do I get 511 instead of 777?


By : furanshisuka
Date : March 29 2020, 07:55 AM
fixed the issue. Will look into that further 0777 is an octal representation.
In other word, 0777 = 7 * (8**2) + 7 * (8**1) + 7 * (8**0)
code :
>>> 0777
511
>>> 7 * (8**2) + 7 * (8**1) + 7 * (8**0)
511

>>> 0777 == 777
False
>>> oct(511)
'0777'
>>> '%o' % 511
'777'
>>> '{:o}'.format(511)
'777'
How to configure REFER call in SIPML5 WebRTC?

How to configure REFER call in SIPML5 WebRTC?


By : cariberecord
Date : March 29 2020, 07:55 AM
should help you out I found it on my own.
SIPML5 does not support call refer but call transfer. So, it uses REFER to transfer call but then, the callee gets disconnected.
Twilio WebRTC client to receive incoming call

Twilio WebRTC client to receive incoming call


By : user2378727
Date : March 29 2020, 07:55 AM
I wish did fix the issue. I've missed the important part of documentation, that can be found here.
What I basically have missed is that when you're calling a WebRTC client, you should prefix the clientID with client: - so the correct request is:
code :
{
  "from": "+1-202-555-0112",
  "to": "client:the_user_id",
  "url": "https://your-server.com/twilio_ml/webhook.xml"
}
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