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[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]Re: [tlug] VoIP help
- Date: Thu, 19 Oct 2017 04:48:34 +0900
- From: Benjamin Kowarsch <trijezdci@example.com>
- Subject: Re: [tlug] VoIP help
- References: <CAKXLc7dNTVfodExfpwwiOsd_kAuHKmjg38HKG1MR9D_VVR+cxw@mail.gmail.com> <B1DA392D-F976-46FF-8AFC-88C45F9A808B@gmail.com> <CAKXLc7dF7CNfqJWdbmD+eN27H8Q-_x6p1kdOA22uenSp=1du8g@mail.gmail.com>
> On Oct 18, 2017, at 22:06, Kalin KOZHUHAROV <me.kalin@example.com> wrote: > > On Wed, Oct 18, 2017 at 7:14 AM, Benjamin Kowarsch <trijezdci@example.com> wrote: >> >> (1) replace that fritzbox with a solidly engineered open source IP-PBX >> solution: >> >> https://freeswitch.org/ >> >> (a) use the fritzbox solely as DECT base station >> >> (b) get a SIP driven DECT base station to hook up your DECT phones >> >> [freeswitch]---SIP---[SIP/DECT basestation]---DECT---[DECT phone] >> >> (c) replace the DECT phones with IP phones >> >> > As far as I can see this is more or less choice 2 above :-D > I have done some preliminary setup of Asterisk on a GL-iNet AR150 > running OpenWRT, may test that. > My asterisk skills are a bit rusty though... > I hope that `directmedia=yes` in sip.conf will be enough to not run > the RTP traffic through the router, just SIP. Freeswitch isn't Asterisk. Asterisk is pretty bad. Freeswitch is solid. If you only have three phones you needn't be concerned about the load of the audio traffic on your router. Three concurrent calls using G.711 encoded audio would be 384 kbit/sec. The only kind of scenario where you might run into load issues would be if you ran Freeswitch directly on your router/firewall (for example if you use pfSense which has a Freeswitch plugin) on an underpowered embedded device while also doing transcoding from or to a computationally expensive codec like G.729. If any part of your setup is behind NAT, you'd be well advised not to split audio and signalling because this can significantly complicate matters. >> (2) replace that fritzbox with a not so well engineered somewhat shaky but >> open source IP-PBX: >> >> Asterisk >> >> then use the Osmocom project's open source DECT stack to drive your DECT >> phones >> >> http://dect.osmocom.org/ >> > Argh! I spend hours looking for something like this and found nothing! > But, looking at the hardware support, I am not sure I can find a > suitable solution (cheap box) that has this chipset. The best option would be to use a dedicated SIP driven DECT base station (or use the Fritzbox as such). This way you won't have to bother with Asterisk either which is inferior to Freeswitch. Note that DECT is the European equivalent of PHS (though not compatible). Only that it was never used to deliver public phone service but strictly as a cordless technology. As such it is a bit dated nowadays. This is not to say DECT and PHS are bad. In fact they are vastly superior to VoIP over WiFi and even superior to GSM and 3G telephony (with the exception of the dynamic channel allocation mode which entered the 3G standard only at the behest of the Chinese mobile phone operators because their GSM services could not compete with the 200+ rogue PHS networks that had sprung up all over China). However, since DECT and PHS have fallen out of fashion, naturally there isn't much interface hardware around now. Switzerland will switch off its ISDN network at the end of this year. This is the first sign of things to come: even the days of ISDN supporting hardware like the Fritzbox are numbered. >> You can run any of this on a small embedded device like the ones from >> >> http://www.pcengines.ch/ >> > Those look nice, I'll get one, may be not for this particular project, > but I am tiny-linux-router ใใงใ :-D You can run just about any OS on those devices. regards benjamin
- References:
- [tlug] VoIP help
- From: Kalin KOZHUHAROV
- Re: [tlug] VoIP help
- From: Benjamin Kowarsch
- Re: [tlug] VoIP help
- From: Kalin KOZHUHAROV
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