Project Requirements¶

{warning} This section is incomplete.

Relevant Problems That Exist Today¶

• Codec negotiation (both with Asterisk, and across a bridge)
  • Support for audio codecs with attributes (SILK)
  • Support for video codecs with attributes
• Limitation on the number of codecs Asterisk can support
• Translation paths are audio specific (with no concept of attributes)
• There is no way to renegotiate codecs after a call is up
• Conferencing is limited to 8 kHz
• There is no way to easily get Asterisk to pass through a media type that it does not understand (proprietary data)
• Once Asterisk supports codecs with attributes, users will need to be able to specify codecs with attributes
• Asterisk is not able to handle a call with more than one audio/video/text stream (only one stream per type).
• Asterisk has no RTCP support relevant to audio and video synchronization
• Asterisk does not support Gtalk video

Phase 1 Requirements¶

Rework media representation completely across all of Asterisk, while maintaining existing functionality. Only add functionality that is required to exercise what has been done.

• Design a new way to represent codecs
  • translation interface
  • capabilities
  • ast_frame handling
  • Initial call setup codec negotiation
  • everything that touches media ...
• Exercise what we have done so far
  • Add support for SILK (and its attributes)
  • Add support for H.264 attributes
• Custom format definitions with attributes (for setting preferences)

Phase Later Requirements¶

• Codec re-negotiation
• Improved conferencing (dynamic sample rate support)
• Video transcoding (an implementation that proves it works)
• A&V sync in RTCP (research required)
• GTalk Video Support
• Support for unknown media types for pass-through (research required)
• Support for more than one stream of the same type (audio/video/text)