-
Notifications
You must be signed in to change notification settings - Fork 3
Ikran Questions
Cisco Systems has open sourced and contributed the code from their enterprise SIP stack to the Ikran project.
The code is released under the Mozilla Public License
The design goal with Ikran's SIP stack is interoperability. To best meet the design goal, the Ikran SIP stack will support a subset of the SIP user agent methods.
Supported SIP Methods:
- INVITE
- REGISTER
- BYE
- ACK
- CANCEL
- PUBLISH
- SUBSCRIBE
- REFER
- INFO
- OPTIONS
- UPDATE
More information on SIP can be found within RFC 3261 for the UPDATE method, more information can be found in RFC 3311.
###What features are available in the SIP stack Ikran has a minimal set of basic phone features. A possible design goal here would be to provide a means to extend this basic feature set via the JavaScript API such that new or custom functionality could be added without needing to modify the underlying SIP stack codebase.
Supported Features:
- Line Registration
- Register Keep-Alive
- Transport Protocols
- Basic Call
- Hold/Resume
- Attended Transfer
- Blind Transfer
- 3-Way Call (locally mixed)
- Call Forward (all, busy)
- Message Waiting Indication
- Roster
- KPML (Key Press Markup Language) Digit Dialing
- Out-of-band DTMF
- Music On Hold
###What IPPBXs will Ikran inter-operate with?
- Cisco Unified Communications Manager
- Asterisk PBX
- More coming soon
###What audio/video codecs will Ikran support? Ikran will leverage the codecs from the WebRTC effort. This means initially Ikran will support the the iLBC audio codec and the VP8 video codec. As the IETF web-rtc codec specifications solidify, there will likely be additional audio codec support for: PCMU/PCMA, DTMF telephony tones and signals and Opus.