Returning to an old thread, again:
I reproduced the reported audio delay issue with ExpressTalk and FreeSWITCH.
Phoner Lite appears to be good, but it needs additional work on its accessibility for screen reader users.
As earlier noted, MicroSIP doesn't support IPv6. It does, however, support the Opus codec. There's what appears to be a drag and drop interface for codec selection, which I wasn't able to use, so I configured the codec preference order by editing the configuration file.
I haven't found an accessible means of issuing DTMF tones in MicroSIP after a call has started. As noted in the documentation, you can append a sequence of commas and digits to the number to be called, resulting in pauses (the commas) and DTMF tones (the digits) being issued as soon as the connection is made. However, this does not satisfy the need to use the dial pad while the call is in progress.
I hope these notes are useful. Comments and questions are welcome.
-----Original Message-----
From: Scott Granados [mailto:scott@granados-llc.net]
Sent: Wednesday, November 1, 2017 3:36 PM
To: Jason White
On Nov 1, 2017, at 11:30 AM, Jason White
wrote: Scott Granados
wrote: That’s a tough one. IPv6 isn’t that common with SIP products yet. We run in to this a lot on our corporate systems. I use XLite and Bria but I don’t think these have the specific features you need.
SIP is also a good use case for IPv6, as it has real problems with NAT. My home network and virtual private server both have IPv6 natively, as do several of the SIP destinations (friends and family, mostly) taht I call regularly. I can configure a Stun server and work around the NAT issues, but IPv6 is the simpler solution - also supported by FreeSWITCH and Asterisk at the server end.