I want to use livekit to send incoming calls from ...
# ask-ai
p
I want to use livekit to send incoming calls from PSTN through SIP to my custom made reactnative app with the SDK. I know LiveKit has a SIP to WebRTC bridge to accept incoming SIP connections but given that I'm focused on australia, having a look onnline it looks the the reigons for SIP pinning by livekit don't contain australia? I'm concerned this wil cause latency issues, given it says LiveKit rooms aer hosted around the world including australia. Am I correct in my concerns? Is self-hosing my only solution?
t
Thanks for reaching out, Sunny. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.
Thank you for reaching out! I wanted to let you know that LiveKit doesn’t currently support SIP region pinning in Australia. While you can definitely host LiveKit rooms in Australia for general services, SIP-specific region pinning is available in regions like the EU, India, South America, and the US at the moment. If latency is something you’re worried about, self-hosting could be a great option to help ensure the best performance in Australia. Also, it might be worth contacting LiveKit directly to see if they have plans to support SIP region pinning in Australia down the road. We’re here to help you navigate this and find the best solution for your needs! Sources: Docs::Home | SIP cloud and region pinning | LiveKit Docs
p
If I selfhost the SIP bridge do I need to self host the rooms themselves as well? I noticed in readme for the bridge it says tis is config file
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# required fields
api_key: livekit server api key. LIVEKIT_API_KEY env can be used instead
api_secret: livekit server api secret. LIVEKIT_API_SECRET env can be used instead
ws_url: livekit server websocket url. LIVEKIT_WS_URL env can be used instead
redis:
  address: must be the same redis address used by your livekit server
  username: redis username
  password: redis password
  db: redis db
the redis address being same as livekit server makes mes assume I need to selfhost both or can i just selfhost bridge?
t
I'm glad you asked about the LiveKit SIP bridge! You can definitely self-host it on its own, but just a heads-up—it needs to connect to a LiveKit server and share some infrastructure like Redis. So, if you're going the self-hosting route for the SIP bridge, you'll want to also self-host the LiveKit server to keep everything running smoothly and compatible. Looking at the configuration file you mentioned, it clearly shows this connection since the SIP bridge uses the same Redis instance and WebSocket URL as the LiveKit server. If you have any more questions or need a hand with deployment or setup, please feel free to reach out—I’m here to help! Sources: Docs::Home | SIP overview | LiveKit Docs | Self-hosted SIP server | LiveKit Docs | test/integration/sip_test.go
r
yes self host but its not that hard
t
You're absolutely right—self-hosting the SIP bridge and LiveKit server is definitely doable, especially with the handy Docker Compose setup or the native deployment options we provide. If you ever need a hand with configuration or deployment, please don't hesitate to reach out—I’m here to help!
p
@rhythmic-furniture-71949 thanks for jumping in, do you have any resources you could point me to? bit concerned about how self-hosting will hold up in a production environment also u reckon if I selfhost SIP, then rest has to be self hosted as well?
t
I don't have the answer you're looking for. You could also try asking your question: • in one of the other Slack channels or • to https://deepwiki.com/livekit/livekit_composite which is trained on all LiveKit source code If you find the answer, please post it here to help others!
r
@polite-school-16734 it scales quite well. does cost a little more to self host but at scale def not download livekit - sip github and livekit server. self host both on the same vm. then u can run agents anywhere. if you get a secure wss://(not needed fordevelopment, u can use ws://) then it is jsut as secure as cloud on prod note you dont get noise cancelling and telephony, as those come in built into the cloud and not the server. but, it makes sip super easy to work with. the lk cli commands are super outdated on the docs so u gotta search for them urself when using them. note: lk cli is also open sourced so u can go and download that then use cursor to give u the right commands. also, deployment is nontrivial; sometimes it takes a while to spin up. the agent can be hosted elsewhere and can point to ur VM