RFC MNCC to SIP bridge

[Holger Freyther]

Hi,

I am looking into building a very simple MNCC to SIP bridge to be used with the NITB (and later the CSCN). This will be based on what was learned from adding the RTP bridge which means the audio will not flow through the bridge (eventually later there will be a UDP proxy for NAT traversal)

The primary design points are:

	* Being able to select TCH/F or TCH/H
	* Decide on the codec very late
	* Support for AMR parameters through MNCC


I just look at MO and MT Call establishment from a high-level point of view:


MO-Call:

	* NITB will send the SETUP indication
	* MNCC GW will do screening and decide if to proceed
	* MNCC GW will make TCH/F or TCH/H decision and ask for a source IP, source port
	* Based on bearer caps (to be handled by MNCC GW?) and TCH/F, TCH/H MNCC GW can offer a SDP file with multiple codecs to the PBX.
	...
	* PBX will decide on a codec (ringing or 200 connect)
	* MNCC GW will ask NITB to reconfigure and audio will flow
	* (TODO: check ringtone, check default, check codec change)



MT-Call:

	* MNCC GW will be presented with a list of codecs already
	* Depending on that TCH/F or TCH/H can/must be chosen (e.g. for AMR even the codec rate can be in the SDP file that decides which TCH/F or TCH/H to use)
	* Can decide bearer caps once paging has succeeded and call is progressing
	* Sets audio codec and waits for call to connect


Handover support:

	* IP/port (and then SSRC and timestamp in RTP) will change
	* MNCC GW could try SIP re-invite with changed parameters
	* MMCC GW could hope PBX implements RTP annex and re-learns the connection



Do you have comments or remarks? The above will lead to a version update, probably a dedicated assignment command in MNCC, and separating socket creation and codec change. Anyone wants to have a go at that?


have a nice weekend

	holger