Sippy B2BUA
A high-performance RFC3261-compliant SIP stack and Back-to-back User Agent (B2BUA) server for call control and accounting.
What is Sippy B2BUA?
Sippy B2BUA is an open-source, RFC3261-compliant Session Initiation Protocol (SIP) stack and Back-to-back User Agent (B2BUA) server. It maintains full call state to enable advanced telephony functions like call accounting, pre-paid billing, and failover routing that are not possible with standard SIP proxies. Unlike PBX solutions, it does not process media, avoiding added latency or jitter.
Telephony engineers, VoIP service providers, and developers building or integrating SIP-based call control systems, especially those needing accurate billing, WebRTC gateways, or high-session capacity.
Developers choose Sippy B2BUA for its high performance (thousands of simultaneous calls), flexibility in media handling via RTPproxy, and ability to implement complex call routing and accounting without the overhead of media processing.
Overview
Sippy B2BUA is a RFC3261-compliant Session Initiation Protocol (SIP) stack and Back-to-back user agent (B2BUA) server software.
Use Cases
Best For
- Building high-capacity WebRTC to SIP gateways
- Implementing accurate call accounting and pre-paid billing systems
- Creating failover call routing solutions for VoIP networks
- Integrating with OpenSIPS or SIP Express Router for complete softswitches
- Handling RADIUS AAA for Cisco-compatible telephony infrastructure
- Developing SIP-based services requiring real-time call control and CDR generation
Not Ideal For
- Projects requiring integrated media processing or PBX features like voicemail and IVR
- Simple SIP proxying where only transaction-level state management is sufficient
- Teams without telephony expertise seeking out-of-the-box, easy-to-configure solutions
- Environments prioritizing minimal external dependencies, as media handling requires separate components like RTPproxy
Pros & Cons
Pros
Supports 5,000-10,000 simultaneous sessions and 150-400 call setups per second, enabling scalable telephony deployments as documented in the features.
Maintains complete call state to generate precise CDRs and implement pre-paid billing, a key advantage over SIP proxies highlighted in the description.
Offers RFC7118-compatible secure websocket support for building high-capacity WebRTC to SIP gateways, as noted in the news section.
Optional use of Sippy RTPproxy allows media relaying without introducing latency, keeping the B2BUA efficient for call control, as mentioned in the features.
Cons
Does not handle media relaying or transcoding, requiring external components like RTPproxy for such functionality, unlike PBX solutions such as Asterisk.
Requires configuration with external systems and deep understanding of SIP/B2BUA concepts, which can be challenging for newcomers, as seen in the running example and dependency on other software.
Focuses on core B2BUA functions; advanced services like billing or WebRTC support need custom implementation or integration, increasing initial development effort.
Frequently Asked Questions
Related Projects
FlexisipLinphone.org mirror for flexisip (git://git.linphone.org/flexisip.git)
KamailioOpen source SIP server widely deployed by carriers and providers. Formerly known as OpenSER
OpenSIPSOpen source SIP server, tracing its roots in OpenSER (presently Kamailio)
RoutrLightweight SIP proxy, location server, and registrar written in Node.js
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