How does Livekit provides the services to Indian M...
# ask-ai
r
How does Livekit provides the services to Indian Market knowing that legally SIP URI integration in India is not done by anyone and no telecom operators provide the SIP URI integration. Everyone works using Websocket. Is there any setup for this available in Livekit to work within Indian Region for Telephony?
t
Thanks for reaching out, Ninad. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.
I'm glad to share that LiveKit supports telephony integration in the Indian region using SIP trunking with region pinning, which helps ensure everything stays compliant with local regulations. Here are some important details to help you get started: • Region Pinning for India: You can easily configure your SIP trunking provider to use a region-based endpoint like
{sip_subdomain}.india.sip.livekit.cloud
. This way, your calls will be routed through LiveKit’s India region smoothly. • WebSocket Usage: Although SIP uses standard signaling, LiveKit’s platform uses WebSocket connections for participant control and room management. This is consistent across all regions and works perfectly alongside SIP telephony. • Telephony Integration: LiveKit supports both inbound and outbound telephony calls, which can be bridged into LiveKit rooms. We also support handy features like DTMF, SIP REFER, and authentication to make your experience seamless. To get everything set up, just configure your SIP trunk with the
.india.sip.livekit.cloud
endpoint. If you want to dive deeper or need more guidance, these resources are great to check out: • SIP Cloud and Region Pinning • SIP Trunk Setup Guide Sources: SIP cloud and region pinning | LiveKit Docs | lib/src/support/websocket/web.dart | Agents telephony integration | LiveKit Docs | Integrating with Telephony | LiveKit Docs | README.md
r
The one which I am working with in India wants the websocket to be configured for Inbound calls, now Livekit on inbound uses SIP URI then how can I get the wss link to put in as websocket url for inbound
t
Thanks for reaching out! I wanted to share that LiveKit doesn’t support using WebSocket (wss) for inbound SIP calls just yet. Inbound calls are managed through SIP over UDP, TCP, or TLS instead of WebSocket. To get your inbound calls working smoothly, you’ll want to set up your SIP trunk with your provider and point it to your LiveKit SIP endpoint. You can find this endpoint from your LiveKit SIP URI in your Project settings. For example, if your SIP URI is
sip:vjnxecm0tjk.sip.livekit.cloud
, then your SIP endpoint would be
vjnxecm0tjk.sip.livekit.cloud
. If securing the connection is important to you, you can use SIP over TLS by adding
;transport=tls
to your SIP URI or by setting the transport port to 5061 and enabling TLS in your provider’s settings. Just keep in mind, this is SIP-level TLS, not WebSocket-based communication. I understand this might feel a bit technical, so please don’t hesitate to ask if you’d like help with the SIP configuration or if you want to explore other options. I’m here to support you! Sources: components/esp_webrtc/impl/apprtc_signal/signal_default.c | SIP trunk setup | LiveKit Docs | Secure trunking | LiveKit Docs
r
So you mean in that case, my wss url would become wss://vjnxecm0tjk.sip.livekit.cloud?
t
Hi there! I wanted to let you know that the URL
<wss://vjnxecm0tjk.sip.livekit.cloud>
isn’t valid for SIP communication. LiveKit doesn’t support WebSocket (wss) as a transport protocol for SIP calls. Instead, you’ll want to use the standard SIP transport protocols like UDP, TCP, or TLS. For inbound calls, your SIP provider should route calls to your SIP endpoint, which in this case would be
vjnxecm0tjk.sip.livekit.cloud
. If you’d like to secure the connection, you can use SIP over TLS by adding
;transport=tls
to your SIP URI or by configuring your provider to use port 5061 for TLS. Please feel free to reach out if you’d like me to clarify anything further or help you set up your SIP trunk. I’m here to help!
r
help me with that setup, If in case I want to go with tls transport protocol then what would be my setup
t
I'd be happy to help you configure SIP over TLS for inbound calls in LiveKit! Here's a simple step-by-step guide to get you started: 1. First, grab your SIP URI from your LiveKit project settings. It usually looks something like
sip:vjnxecm0tjk.sip.livekit.cloud
. 2. Next, share this SIP URI with your SIP trunking provider and make sure they route calls to that endpoint. 3. To turn on TLS, just add
;transport=tls
to the end of your SIP URI. For example:
sip:vjnxecm0tjk.sip.livekit.cloud;transport=tls
. 4. Double-check that your SIP provider is set to use port 5061, which is the standard port for SIP over TLS. 5. Also, please confirm your SIP provider supports TLS and has the right certificates in place to keep your connection secure. If you have any questions or need a hand with any of these steps, feel free to reach out. I'm here to help!
r
SIP provider does not support configuring over sip uri, they only work with websocket and no other provider in india works with SIP URI. everyone works over websocket only
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
give me support email for livekit