I would like to increase the loudness of the calls at the originating point. The mic on the set is no problem, but the earpiece or speaker at the originating end lacks acceptable volume. The volume thru windows is fine and adjustable, but does not allow communication volume adjustment. This problem is only apparent at the originating end.
Rob from Twilio here.
Thanks for the question on controlling volume of a call. Sounds from your description of "Windows" that you are accepting a call and answering with Twilio Client in a Windows-based browser - if I'm incorrect pleased let me know. There is no API control of the volume of a given participant in a call (full reference on modifying live calls here).
If the person calling into the Client user is having difficulty hearing, you may want to look at your microphone settings on your machine. Also make sure that you are using a headset and not your computer's built-in microphone. More tips on troubleshooting Client sound quality are available here.
Related
I'm currently experiencing an intermittent issue with some VOIP WebRTC voice calls.
The symptom is that the outbound audio can sometimes fade in and out and sounds extremely muffled or even disappears momentarily. The 2 audio files reference here show examples or a snippet from a good call and then a bad call, both very close together. The audio is captured server side.
Good quality call - https://s3-eu-west-1.amazonaws.com/audio-samples-mlcl/Good.mp3
Poor quality call - https://s3-eu-west-1.amazonaws.com/audio-samples-mlcl/Poor.mp3
The tech stack is comprised of…
Electron application running on Mac/Windows
Electron wraps Chromium v66
WebRTC used within Chromium
OPUS codec used from client to server.
Wired network connection (stats show no packet loss and Jitter, RTT and delay are all very low)
SRTP used for media between client and TURN server (Coturn)
Cotur
Janus WebRTC Gateway
Freeswitch
These are using high-quality headsets and have been tested with various different manufacturers connecting to the Mac/Windows using USB.
Any ideas/help would be greatly be appreciated.
This might be a result of auto gain control. Try disabling it by passing autoGainControl: false to getUserMedia. In Chrome/electron googAutoGainControl and googAutoGainControl2 might still work.
I'm trying to send a second source of audio (streamed from a bluetooth connected device as Int16 data flow) to a room using twilio-video on iOS.
I already looked at TVIAudioController but found nothing about audioInput.
Checked-out also TVIAudioSink but I don't know if it can do the job.
Anyone can help ?
Twilio developer evangelist here.
I'm not an expert in this at all, but since no-one else has helped yet, I can try to point you in the right direction. It sounds like you're trying to hack around the limitation of using one microphone on an iOS device at a time. Good luck!
In order to send another audio track, you need to create a TVILocalAudioTrack. Normally this is initialised using the track method, which picks the audio input itself. It's not recommended to use the regular initializer, but that is probably worth exploring.
Once you have your TVILocalAudioTrack you need to add it to your room Participant using addAudioTrack.
Let me know if that helps at all.
I believe the answer is no, but does Twilio provide ability to pause/resume a recording? Use case is recording a call, but pausing recording when collecting sensitive information. From the REST documentation, it doesn't appear to be a supported capability. Thought someone might have found some options for this requirement.
This is possible, though it's not wholly obvious how from the documentation.
You can modify call state using the REST API, as per https://www.twilio.com/docs/api/rest/change-call-state , and we basically use it to tell the call to re-dial to the same agent (presumably this is a call centre?) but with no-record, and then again with record re-activated once we're done.
You end up with two separate recordings for the call, which in our case we download, stitch together, and store back to our storage platform.
Edit:
Having discussed this issue with Twilio support, there's another possibility which allows you to just have a single recording.
Instead of dialling the two ends of the call together, you instead put them both into a conference that's recorded when you initially connect the call. When you want to pause it, using the REST API, you add a new "hold" leg into the conference, then move the two real ends of the call onto a new conference that isn't recorded. When you're done, you move them back again and it's "unpaused". You then only have one recording from the original conference.
None of these is ideal, and apparently they are working on a proper support setup for this (fairly obvious!) requirement, but this should solve it for now.
The Recording Pause & Resume feature is now supported in the Twilio API. Here's a link that gets you started:
https://support.twilio.com/hc/en-us/articles/360010199074-Getting-Started-with-Call-Recording-Controls#pause_resume
I need to capture traffic Information about a live event on the Internet. I need as much details as possible about the traffic. (e.g. number of viewers from a particular region, Devices the viewers are using, type of video stream (e.g. 240p) and so on.
I see that all this information can be captured using the APIs provided on YouTube Live Streaming Event. However, the APIs can only be used by the owner of the channel broadcasting the event. I can set up my own event using YouTube Live Streaming or setting up my own server and gather the required statistics, however it would be better if I have the data from an established source, so the traffic data obtained is good to work upon for major events.
I have already tried speaking to a few of the channel owners with no luck. Is there a way, I could obtain this data? (e.g. Capture traffic on a third party web page like YouTube).
I would say that while it may be theoretically possible to capture live event statistics by scraping the video's webpage, monitoring packets, or reverse-engineering their API, this is against YouTube's Terms of Service:
You agree not to circumvent, disable or otherwise interfere with security-related features of the Service or features that prevent or restrict use or copying of any Content or enforce limitations on use of the Service or the Content therein.
https://www.youtube.com/static?template=terms
When programs such as Skype streams video from a user to another and vice versa, how is that usually accomplished?
Does client A stream to a server, and server sends it to client B?
or does it go directly from client A to B?
Feel free to correct me if i am way off and none of those is correct.
Skype is much more complicated than that, because it is Peer to Peer, meaning that your stream may travel through several other skype clients, acting as several servers. Skype does not have a huge central system for this. Skype always keeps track of multiple places that it can deliver your stream to, so that if one of these places disappear (that Skype client disappears), then it will continue sending through another server/skype-client. This is done so efficiently, that you don't notice the interruption.
Basically , this is how its achieved.
1) encode video / audio using the best compression you can get. Go lossy compression and plenty of aliasing to throw away portions of video and audio which is not usable. Like removing background hiss
2) pack video / audio into packets and put a timestamp on them. The packets are usually datagrams.
3) send packets directly to destination. Use the most appropriate route. You dont have to send all packets the same way. Use many routes if possible. P2P networks often use many routes to the same destination
4) re-encode on the destination. If a packet is too old , throw it away. If packets are lost , dont bother about it since its too late.
5) join the video back and fill in the missing frames the best you can.