Re: VoIP+IAX Program Theory for OM
Aside from those who run their own asterisk server, remember those who just use an iax itsp! Anyways, I have been working on the core and backend library (iaxclient) of mokoiax. You can read about a release I did of a command line based version for the openmoko (installable through ipkg) on http://bkruse.com. I am also looking for developers. To give a quick idea, I started working on a gtk version (mokoiax) designed specifically for the moko when I then realized why not tie it into the dialer application itself? Anyways, let me know your thoughts. Let me know if you want to help by emailing me at this address or [EMAIL PROTECTED] ( this is in no way affiliated with digium, I just happen to work there :) ) Brandon Kruse (bkruse) On Feb 22, 2008, at 8:13 AM, Jonathan Spooner <[EMAIL PROTECTED] > wrote: I too run my own asterisk server. I'd think IAX support on an OM client would be critical (at least untill were all on ipv6). The only reason you'd run a voip client on OM is so you can roam from voip to GSM with a preference to VOIP when wifi is available so supporting IAX would make this as painful as possible. Excellent idea! I'd be happy to help with anything other than coding once I get a freerunner. Regards, JonS Kyle Bassett wrote: Thanks for all the input! To clarify: I have already set this this system up using linux/win/mac IAX clients and it works great. Reliability is very high (no failures within the 4 months I've had it up) with my dedicated asterisk server running off my DSL connection (QoS on with a linux router). If the asterisk server cannot reach me via a VoIP connection, it fallsback to calling my cell phone number. If the asterisk box fails for whatever reason, my VoIP provider has a fallback number to dial as well. The asterisk server just has a VoIP account for inbound and outbound calls, no analog lines are connected. The cost benefit here would be the ability to accept a lower plan from your cell provider (possibly data-only when 3G is available?), or even use a prepaid service with the smartphone. I am currently using a per- minute VoIP/POTS termination plan with no monthly fee, which works out to be much cheaper with the lower cellular plan. I have not wrote the application as of yet, I wanted to gauge interest for a project like this. If I do write this application, I would like to implement encryption along the way. In addition, I would set up an asterisk box at our business location for testing within a larger userbase. The reason I prefer to use a full asterisk system is the ability to integrate it within our business. I prefer IAX over SIP because it is NAT routeable, whereas SIP has many issues with firewall traversal. In reality, the client should support both. Keep it coming! :-) -Kyle --- - ___ OpenMoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community -- Jonathan Spooner Nationwilcox Systems Ltd Tel: 0121 3544345 ___ OpenMoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community ___ OpenMoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: VoIP+IAX Program Theory for OM
I too run my own asterisk server. I'd think IAX support on an OM client would be critical (at least untill were all on ipv6). The only reason you'd run a voip client on OM is so you can roam from voip to GSM with a preference to VOIP when wifi is available so supporting IAX would make this as painful as possible. Excellent idea! I'd be happy to help with anything other than coding once I get a freerunner. Regards, JonS Kyle Bassett wrote: Thanks for all the input! To clarify: I have already set this this system up using linux/win/mac IAX clients and it works great. Reliability is very high (no failures within the 4 months I've had it up) with my dedicated asterisk server running off my DSL connection (QoS on with a linux router). If the asterisk server cannot reach me via a VoIP connection, it fallsback to calling my cell phone number. If the asterisk box fails for whatever reason, my VoIP provider has a fallback number to dial as well. The asterisk server just has a VoIP account for inbound and outbound calls, no analog lines are connected. The cost benefit here would be the ability to accept a lower plan from your cell provider (possibly data-only when 3G is available?), or even use a prepaid service with the smartphone. I am currently using a per-minute VoIP/POTS termination plan with no monthly fee, which works out to be much cheaper with the lower cellular plan. I have not wrote the application as of yet, I wanted to gauge interest for a project like this. If I do write this application, I would like to implement encryption along the way. In addition, I would set up an asterisk box at our business location for testing within a larger userbase. The reason I prefer to use a full asterisk system is the ability to integrate it within our business. I prefer IAX over SIP because it is NAT routeable, whereas SIP has many issues with firewall traversal. In reality, the client should support both. Keep it coming! :-) -Kyle ___ OpenMoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community -- Jonathan Spooner Nationwilcox Systems Ltd Tel: 0121 3544345 ___ OpenMoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: VoIP+IAX Program Theory for OM
http://bkruse.com Brandon Kruse (bkruse) On Feb 22, 2008, at 2:52 AM, "Sébastien Lorquet" <[EMAIL PROTECTED]> wrote: Hi all, Not really the main subject of the thread, but let me recall that UMA is not possible on OpenMoko, since it requires direct access to the internals of the GSM modem (SIM access and others). http://lists.openmoko.org/pipermail/community/2007-September/010575.html Moreover, at least in France, it's "forbidden" to use 3G data links to transfer VoIP steams. Just because it would be cheaper than voice plans, I guess :) Sebastien ___ OpenMoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community ___ OpenMoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community
Re: VoIP+IAX Program Theory for OM
Hi all, Not really the main subject of the thread, but let me recall that UMA is not possible on OpenMoko, since it requires direct access to the internals of the GSM modem (SIM access and others). http://lists.openmoko.org/pipermail/community/2007-September/010575.html Moreover, at least in France, it's "forbidden" to use 3G data links to transfer VoIP steams. Just because it would be cheaper than voice plans, I guess :) Sebastien ___ OpenMoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community