On Sep 24, 2005, at 9:02 AM, linux-audio-dev- request(a)music.columbia.edu wrote: I'd recommend using RTP as your base protocol, and defining your SPIF or SC3-like payload as an RTP payload format. You'll pick up the entire IETF multimedia protocols for free this way, including RTP MIDI: http://www.cs.berkeley.edu/~lazzaro/rtpmidi/index.html I think when it comes to networking, the writing is on the wall when in comes to packet loss being a part of the environment you need to live in. Most new purchases of computers are for laptop computers, most of those users want to use WiFi as their network, and the Internet layer sees 1-2% packet loss on WiFi. Also, we live in an era where people want to run LAN apps on the WAN and WAN apps on the LAN, and packet loss is also an unavoidable part of the WAN Internet experience. Finally, modern applications want to use link-local Internet multicast. RTP was built for letting payload formats handle packet losses in a way that makes sense for the media type -- RTP MIDI is an extreme example of this, but the audio and video payload formats are loss tolerant in more subtle ways. RTP is also multi-cast compatible. Finally, with RTP there's a standardized way to use RTSP and SIP for your session management if you wish, or if you prefer, you can just build RTP into whatever session manager you have committed to (like jack). --- John Lazzaro http://www.cs.berkeley.edu/~lazzaro lazzaro [at] cs [dot] berkeley [dot] edu ---