OsMux + MGW + SIP

Hi community! I'd like here to set out a few issues that we have identified for going forward with the rhizomatica autonomous networks and the osmocom split stack. Some of this has been very briefly discussed between myself and Pablo, and I bring it to the mailing list so hopefully there will be feedback,  ideas and suggestions :) I am cherry picking from previous emails here, (mostly mine) please excuse any repetition. I hope others will post their previous comments on this thread, either verbatim of edited to be relevant to the current discussion. 1) This is a question that relates to voip traffic in and out of the autonomous communities over satellite. Currently this is what we HAVE to have in the local village to support autonomous operation): * BSC * MSC * HLR * Freeswitch (freeswitch is our call control, call routing, billing etc) We can't have any essential backhaul running over the satellite , like A for example as we would loose all functionality when the sat link is down, and we'd burn expensive bandwidth! One option to use osmux from autonomous community (village) is to teach osmo-mgw to speak osmux, then we have a co-located bsc/msc/mgw in the village, and the media transport is osmux to another mgw in our datacentre that is demuxing and converting to RTP suitable to be forwarded to our upstream VoIP provider. The question here is what is signalling the MGW in the data centre? Maybe another freeswitch could signal the osmoMGW via some kind of SIP<->MGCP translator. Or we teach the MGW to speak SIP? Another option that I want to put on the table and see what people say is to look at implementing osmux as a codec in freeswitch. I don't know what that would mean in terms of effort. I don't know if the osmux code can be abstracted, if that is the right term, into it's own external includible library that could then be used to build a freeswitch codec. I looked some implementations of AMR codec for freeswitch, and really it looks like a boiler plate codec with registration, setup and then it calls the encoding and decoding functions in the opencore amr library. Maybe, we can do the same with osmux? Then we could have two freeswitchs signalling and transcoding to/from osmux? -  then we are not really working on osmo solution and others in the osmo community might prefer to see osmux fully implemented in MGW before anything else? Here are a number of related tickets from Harald on osmux integration: http://osmocom.org/issues/1683 and http://osmocom.org/issues/2832 as well as https://osmocom.org/issues/2909 I think though, that in 1683, we are stalled by https://osmocom.org/issues/2391 for the split setup. 2) Another question relates to this proposal of a media gateway less mode:  https://osmocom.org/issues/3142 I think this suits us, because really our call signalling is all happening in freeswitch and we would prefer I think not to have the media gateway at all most of the time. and in fact for local Mobile to Mobile calls on the same BTS, we would in fact prefer the RTP be local on the sysmobts, or indeed BTS<->BTS! Freeswitch has a "bypass media" parameter, in fact you can even activate this at call processing time before bridging the call, depending on whether it makes sense in terms of direct connection being possible and the lack of transcoding. There's also a "bypass media after bridge" parameter that is automatically using SIP (re)-INVITES to switch the RTP stream from passing through freeswitch or going directly from end point to end point.  Using "bypass media" in our profile works nicely; of course, as all it is doing is using the IP address(es) of the osmo-bts in the SDP, so the rtp stream loops on lo in the sysmobts. It would help with something that I mentioned at Osmodevcon, Harald, which was that in some cases we might like to avoid having our RTP go over the (sometimes variable quality) Wifi links between the BTS and the BSC. A problem is that if you use "bypass media" you have no early media and no audible ringing signal, you don't get any audio stream until the B leg picks up. This is what "bypass media after bridge" is for. Unfortunately, osmo-nitb (at least, and I don't believe anything has been done in osmo-bsc related) plus LCR or osmo-sip-connector combinations do not react correctly to the SIP reINVITES, you end up with multiple calls. So all that needs to be fixed / implemented. I noticed mention of implementation of MNCC_RTP_MODIFY on fairwaves branches of OpenBSC and LCR. I've compiled and tested this, but as far as I can see it is still not working, as issuing a SIP reINVITE from freeswitch is still setting up new calls, not just changing the RTP streams. Maybe I also need a fairwaves osmo-bts. So this is something I think I would like to see: * Full support for SIP reinvite in osmo-sip-connector which would then send a MNCC_RTP_MODIFY which ends up sending (I think it's called a IPAC_CRCX?) to osmo-bts which will then switch the stream endpoints. This I believe is also necessary for handover to function with an MNCC socket setup and we don't currently have it, not even in the osmo-nitb. * Implement a no media gateway mode in osmo-msc and have the MSC control the media stream using SIP via MNCC/osmo-sip-connector instead of controlling the MGW using MGCP. K/