LiveKit Community

can you share more? what timeout error are you seeing?

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<@U01KZ61GV5Z> if u try to connect to playground via safari ios (tested on 15.2, 15.3) or safari macOS (12.2) you will get Error: could not connect after timeout.

But in safari tech preview everything is ok, as it is in other browser. Second screenshot it is the same timeout error from my server. 

It's just can't establish webrtc connection to the server from safari. 

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<@U01KZ61GV5Z> Here you can see what happens when you try to connect to playground from safari. 

<@U02RL8CR6UC> I'm unable to reproduce this. It's working for me on both iOS and Mac 15.2. are there anything notable about your network?

That sounds like the basic approach. You’ll probably also want to implement web/push notifications in case the callee is not currently active. Btw Agora (another webrtc service) supports invites in their SDK if you want to check how they structured it <https://docs.agora.io/en/Real-time-Messaging/rtm_invite_web?platform=Web>

Thanks for the reply, awesome.  Will definitely take a look at their approach.

<https://github.com/sze-plusplusplus/projektmunka|https://github.com/sze-plusplusplus/projektmunka>

It should be there: <https://docs.livekit.io/client-sdk-js/modules.html#createlocaltracks>

<@U02SG8NA0DU> <@U01KZ61GV5Z> SDK changed in 0.15 it's LocalParticipant.createTracks now
<https://github.com/livekit/client-sdk-js/releases/tag/v0.15.0>

the sdk docs need to be updated

...and I think the typescript definitions are still missing, right?

<@U02HN8GPUBY> we've since restored the APIs :slightly_smiling_face: Since many were dependent on them, we didn't want to break a bunch of people's apps.

```const tracks = await createLocalTracks({
  audio: true,
  video: true,
});```

it's exported via:
```export * from './room/track/create';```

ok I will try, but why isn't it in the general exports, as its a very basic function?

BTW: I find liveKit extremely promising and I am very exited to evaluate it further - its a great job you guys did so far, my compliments!!

thanks! On the topic of exports, `export *`  exports all of the symbols from another file, so you should be able to `import { createLocalTracks } from 'livekit-client'`
If you do get errors about that, it might have been a version issue. (in v0.15, we removed the top level create methods), they were reintroduced back in the patch releases

```export { setLogLevel, Room, RoomState, DataPacket_Kind, ConnectionQuality, Participant, RemoteParticipant, LocalParticipant, LocalAudioTrack, LocalVideoTrack, LocalTrack, LocalTrackPublication, RemoteAudioTrack, RemoteVideoTrack, RemoteTrackPublication, TrackPublication, VideoQuality, };```

Have you seen WebTransport ? It looks interesting tbh

it seems that they took a lot of the lessons from webrtc, and perhaps it could become the layer that can power it all

Yes why not, I think it is also very good for video games since there is no easy solution to use an unreliable channel.
Learning WebRTC is something less of an option for game developers

is that a personal project or for a company?

I haven't seen any company using WebRTC for their browser based games

WebRTC is used for videogame streaming so heavy rendering can be done in powerful servers

Hey Clark, I’ve seen a few hits in search for caddy (can’t link to search results in Slack, I guess :disappointed:). If none of those threads answer your question, someone here can probably help.

I've been running livekit with nginx, caddy and traefik.. it works with all of them

Caddyfile
```{$LIVEKIT_SITE_ADDRESS} {
  tls {$ADMIN_EMAIL}
  reverse_proxy /rtc livekit:7880 {
    header_up Host {host}
    header_up X-Real-IP {remote}
    header_up X-Forwarded-For {remote}
    header_up X-Fordwarded-Proto {scheme}
  }
}```
this assumes you have an environment variable LIVEKIT_SITE_ADDRESS=<http://webrtc.yourdomain.com|webrtc.yourdomain.com> (or whatever your public address is) and ADMIN_EMAIL=<mailto:your@email.com|your@email.com>

that is for getting let's encrypt certificates

if you use docker-compose:
```caddy:
  restart: always
  image: caddy:2.4.6-alpine
  volumes:
     - ./Caddyfile::/etc/caddy/Caddyfile
  ports:
    - 80:80
    - 443:443
  environment:
    - LIVEKIT_SITE_ADDRESS=<http://webrtc.yourdomain.com|webrtc.yourdomain.com>
    - ADMIN_EMAIL=your@email.com```
something like that

Awesome thanks very much everyone! I'm looking at using docker compose so that helps. Want this to be the first compose file I make!

<@U02SQNHPNHG> no problem, let me know if you have trouble with it :wink:

<@U02SUCZ6TTN> please stop spamming all the channels with the same messages

in general it's probably a better idea to use your own DB and services for user data and only use user_id to link it.

otherwise you're relying on livekit to be your db

the identity is supposed to be the user_id.  In fact, if a participant joins a room with an identity that is already being used by another participant in that room, it will disconnect the existing participant, as it's considered the same user

btw, you can set participant's metadata when creating the token, and also update users metadata with the server api

you could use <https://docs.livekit.io/guides/room/data#participant-metadata|participant's metadata field> to attach application specific metadata. As Jonathan said, it's not supposed to be a persistent store, but could be used to deliver participant-specific state to other clients.

<@U02BD5MV5NX>,  I have my db,  what am trying to achieved is  that, in a channel  different user will have different roles, which will determine what they can do . The main  aim is something  i can use to identify the remoteParticipants  from  my db.   The metaData as mentioned by <@U01KZ61GV5Z>  will be enough.

You mean, different providers? Probably more work than it’s worth. /shrug

i guess i’m just trying to understand webrtc basics…  webrtc doesn’t have a standard server interface? like you have wss://[host]:7880 … but other server would be different?

Oh, right, so it depends on the needs of your application. But in most cases, you’d probably also have a more traditional web/application server which manages things like your database/persistent storage, auxiliary APIs, user auth, etc.

This could also, of course, be “serverless” — you could use Firebase/Supabase, for example, and your backend would be a collection of cloud functions.

as a specific example… the ant media server seems to have more the recording capability that i could use. but can i use the livekit flutter client to connect?

sorry these are noob questions i’m sure

Ah, ok, that’s different then. I haven’t really looked into the kind of interface(s) Ant supports, but you’d most likely have to modify our client to communicate with Ant. What kinds of recording capabilities do you need that aren’t available?

well i’m playing with lots of crazy ideas in my head

My understanding of it is that the webrtc protocols themselves, much like the general browser media APIs, are pretty standard. The part that isn't standard is using an SFU, like LiveKit/Jitsi/etc.. So, even though the actual way the video is being sent across the wire is standardized, the way you tell the browser how to connect to other clients (the signaling) is different. You could probably get some things to mix together, but you'd need have translation layers for the signaling side of things.

We noticed that the SDK was tagged with 0.8.0 and released on maven - so I guess it's just the github "release" that is missing?

Yes! It's just missing the GH release + notes. 0.8.0 was cut a couple of days ago

release notes to follow soon: CC: <@U01NZKTNE1H>

<@U02K0H5LQUX> I don't know what livekit-avp is, I can't find any reference on the web, I know ION has ion-avp, which is deprecated.

On livekit you already have room recording functionality, it's <https://github.com/livekit/livekit-recorder|livekit-recorder>, which is basically a headless chrome + gstreamer that records your room (or any url you provide) as mp4 file and optionally uploads it to S3 or Azure blob store.

livekit-recorder uses some built-in templates you can choose from, or you could build your own layout, either from scratch or by editing livekit-recorder/web and deploying it on your own server.

Recording through the browser is a very large overhead
<https://livekit-users.slack.com/archives/C01KVTJH6BX/p1641428171296100>

ion-avp is not deprecated and the author abandoned it because he initially chose the wrong path. At the same time, he wrote that the way with gstreamer is more correct and solves these problems. With headless chrome, the option only seems to be successful, but the reality is very high overhead and low stability.

I tried 2 different ways but didn't get good result. Here you can get info: <https://github.com/livekit/livekit-server/issues/59#issuecomment-955639311>
The main problem is if you capture video then video run very fast similar like watching video in fast forward mood. Same for audio. Still waiting for such solution.

<@U02K0H5LQUX> thanks for the offer! This is a great opportunity for those that are interested in this problem.

are you trying to composite multiple streams into a single file? or saving individual streams? If the former, take a look at <https://github.com/cryptagon/ion-cluster/blob/master/cmd/composite.go|this project>. There are some compositing code with GStreamer.

the reason why we went with a Chrome-based solution is due to the flexibility of creating a rich recording interface. Handling the dynamism of participants entering and leaving is quite tricky with something like GStreamer.

Hi All, Great pack first of all. :slightly_smiling_face: I need some help with deploying LiveKit server in Docker with configuration file. I want to manage the incoming UDP port range. Thanks in advance. :slightly_smiling_face: :slightly_smiling_face:

Hi, see <https://github.com/pion/example-webrtc-applications/tree/master/sip-to-webrtc>

And see <https://github.com/ghettovoice/gosip> for sip server. if need gateway sip-&gt;LiveKit