jart/gosip

gosip

Version: 0.1

About

gosip (pronounced "gossip") is a library that lets you make phone calls using the Go programming language. It provides a full-stack SIP/RTP implementation that's tailored towards making calls over the PSTN through services such as Flowroute.

gosip is most suitable for backend telephony applications. This is especially true for apps that do interesting things with audio. gosip supports DSP out of the box by providing SSE optimised audio mixing, an assembly implementation of the µLaw codec (no other codecs are supported), and a comfort noise generator.

Telephony applications have traditionally been written on top of PBX systems like Asterisk via an extension interface. Frameworks such as Adhearsion have introduced further layers of abstraction to this unwieldy regime. These systems are slow, difficult to administer, and in many cases superfluous to the needs of the telephony app developer. gosip sets you free from Asterisk because you can have everything in a single easy-to-deploy binary.

gosip has excellent support for SRV/NAPTR failover by way of timeouts, ICMP refusal, and 502 Service Unavailable responses. It also supports SIP redirects as well as changing the audio/signalling path mid-call. But most importantly, gosip is lightweight enough that you can actually design your app to use a single process for each phone call (assuming your app is in the audio path.) By taking this non-monolithic approach to design, you can actually build a higher availability global telephony service with zero interruptions during software upgrades.

gosip parses SIP messages using the Ragel finite state machine compiler. Much of the BNF was copied directly from the SIP RFC. This implementation approach offers a fair amount of confidence that the parser is not only fast, but also correct and secure.

gosip is less suitable for PBX, IVR, and VoIP reselling. For these things, you should consider Asterisk, FreeSWITCH, or SER.

RFCs

• RFC3261: SIP
• RFC4566: SDP
• RFC3550: RTP
• RFC4733: RTP DTMF
• RFC3262: SIP 100rel
• RFC3263: SIP: Locating a Server

Example RFCs

• RFC4475: SIP Torture Test Messages
• RFC3665: SIP basic call flows
• RFC3666: SIP PSTN call flows
• RFC6216: Example call flows using SIP security mechanisms