The growing network of broadband connections, paired with the arrival of 4G and now 5G services, has provided a unique opportunity to look beyond the bandwidth savings model for video conferencing, and rather focus on optimizing the cost of media delivery through intelligent systems over standard configurations.

From communication to streaming to interactivity, SARISKA offers a comprehensive tech stack to build everything from tiny applications to endless worlds through simple, fast, and secure APIs and SDKs helping you to get from development to launch in a week.


Jitsi’s architecture allows for dynamic scaling in real time. In addition to being powerful and optimized, JVBs are built to scale, which makes them more dynamic for media transport.

Scaling the audio/video conferencing architecture is made easier with the usage of Jisti Videobridges due to the separation of the signaling layer from the transport layer.

Live Streaming

Sariska lets you also live-stream your video conference. Sariska addresses the challenges of concurrent streaming and latency by using a media server called SRS (Simple Realtime Server), which is a simple, high-efficiency, and real-time video server supporting RTMP/WebRTC/HLS/DASH/HTTP-FLV/SRT protocols.

SRS is used as a real-time media delivery server on top of our video conferencing architecture. This enables you to push your streams through multiple outlets in a single click with simulcast-enabled live streaming and reach your audience across every platform.


If you are using Sariska then media is routed through the Jitsi Video bridge. If you have your own WebRTC set up and you need only stun/turn servers for NAT Traversal. Please check out the details below.

SARISKA also provides DNS-based load balanced turn servers, DNS-based load balancing is a specific type of load balancing that uses the DNS to distribute traffic across several servers. It does this by providing different IP addresses in response to DNS queries which are highly available and available to near your location.

you can fetch usernames and passwords to use with turn servers.

URL: https://api.sariska.io/api/v1/misc/fetch-credentials?exp=<expiry>
Method: GET
Headers:  { Authorization: "Bearer your-token" }

// expiry can be passed as following accepted formats:  2 seconds or 2 minutes or 2 hours or 2 days or 2 weeks or 2 years.

Turn Server

A TURN server is a media relay/proxy that allows peers to exchange UDP or TCP media traffic whenever one or both parties are behind NAT.

with UDP:


with TCP:


Stun Server

The STUN server allows clients to find out their public address, the type of NAT they are behind, and the Internet side port associated by the NAT with a particular local port


✅ Runs on port 80 and 443 to penetrate most restricted firewalls in the world

✅ For the lowest-latency P2P calls and to support users with restricted connectivity

✅ Enterprise-grade reliability (99.99% uptime)

✅ Support STUN

✅ Supports both TCP and UDP

✅ Dynamic routing to the nearest server

✅ Production Ready

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