Hello, Unfortunately Microsip does not provide above features, but you can the following application, it has ability to use IPV6 support: http://www.linphone.org/ 2017-11-01 16:23 GMT+03:00, Bill Dengler <codeofdusk@gmail.com>:
Yeah, and also how about Opus at 48 kHz?
Bill
On Nov 1, 2017, at 12:52 AM, Jason White via Blind-sysadmins <blind-sysadmins@lists.hodgsonfamily.org> wrote:
I'm reviving an old thread here. Are there any SIP clients mentioned in this discussion as offering good screen reader accessibility that support IPv6?
-----Original Message----- From: Blind-sysadmins [mailto:blind-sysadmins-bounces@lists.hodgsonfamily.org] On Behalf Of Chris Nestrud Sent: Saturday, September 23, 2017 4:30 PM To: Blind sysadmins list <blind-sysadmins@lists.hodgsonfamily.org> Subject: Re: [Blind-sysadmins] softphone recommendations
My recommendation changes from Express Talk to Phoner Lite. Phoner Lite has some unlabeled buttons and is not as accessible as Express Talk., but Phoner Lite does not have this strange delay issue.
Using same hardware and two softphones with same codec, looks like it might be an Express Talk issue.
Express Talk:
Dial 9196, immediately start saying numbers quickly, stopping when numbers are heard, 1-12 is heard from echo. The echo application is working and sends back pretty much all data but with this 1.5 second delay.
Free Switch log of Express Talk connection:
2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4352 Set telephone-event payload to 101@8000 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:3056 Set Codec sofia/internal/1000@freeswitch.server PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2017-09-23 14:44:06.997035 [DEBUG] switch_core_codec.c:111 sofia/internal/1000@freeswitch.server Original read codec set to PCMU:0 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4695 Set telephone-event payload to 101@8000 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4754 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 recv payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:6865 AUDIO RTP [sofia/internal/1000@freeswitch.server] 72.14.x.x port 29276 -> 71.238.x.x port 8000 codec: 0 ms: 20 2017-09-23 14:44:06.997035 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7166 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7173 sofia/internal/1000@freeswitch.server Set 2833 dtmf receive payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7196 sofia/internal/1000@freeswitch.server Set rtp dtmf delay to 40 2017-09-23 14:44:06.997035 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@freeswitch.server! 2017-09-23 14:44:06.997035 [DEBUG] switch_channel.c:3473 (sofia/internal/1000@freeswitch.server) Callstate Change RINGING -> EARLY 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:6848 Audio params are unchanged for sofia/internal/1000@freeswitch.server.
Phoner Lite:
Same computer and audio device as Express Talk. I set the codec to G.711 u-Law, 64 kbps which matches Express Talk. It was originally set to Opus, 50kbps.
Dial 9196, immediately start saying numbers quickly, stopping when numbers are heard, 2 is heard from echo. The echo application is working and sends back data with minimal delay.
Free Switch log of Phoner Lite connection:
2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4352 Set telephone-event payload to 101@8000 2017-09-23 15:08:52.917044 [WARNING] switch_core_media.c:4606 Invalid SDP for opus. Don't ask.. but it needs a /2 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:3056 Set Codec sofia/internal/1000@freeswitch.server PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2017-09-23 15:08:52.917044 [DEBUG] switch_core_codec.c:111 sofia/internal/1000@freeswitch.server Original read codec set to PCMU:0 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4695 Set telephone-event payload to 101@8000 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4754 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 recv payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:6865 AUDIO RTP [sofia/internal/1000@freeswitch.server] 72.14.x.x port 23668 -> 71.238.x.x port 5062 codec: 0 ms: 20 2017-09-23 15:08:52.917044 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7166 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7173 sofia/internal/1000@freeswitch.server Set 2833 dtmf receive payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7196 sofia/internal/1000@freeswitch.server Set rtp dtmf delay to 40 2017-09-23 15:08:52.917044 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@freeswitch.server! 2017-09-23 15:08:52.917044 [DEBUG] switch_channel.c:3473 (sofia/internal/1000@freeswitch.server) Callstate Change RINGING -> EARLY 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:6848 Audio params are unchanged for sofia/internal/1000@freeswitch.server.
Chris
I haven't had a chance to dig into this further. That sounds like a logical place to begin investigating. Thanks.
Chris
On Thu, Sep 21, 2017 at 02:29:25PM +0000, Scott Granados wrote:
Sounds like it could be a problem in Jitter buffer management. What do you think? Have you tried adjusting the type and size of buffer negotiated by the PBX?
On Sep 21, 2017, at 8:56 AM, Chris Nestrud <ccn@chrisnestrud.com> wrote:
Express Talk has a strange issue for me where several seconds of delay will build up after about an hour of a SIP call being connected. I haven't had this delay issue when using Bria on iOS. Express Talk is very accessible. Let me know if you try it and have
On Thu, Sep 21, 2017 at 09:50:13AM -0500, Chris Nestrud wrote: the delay issue.
Chris
On Tue, Sep 19, 2017 at 08:21:34PM -0500, Chris Nestrud wrote:
Express Talk from NCH looks accessible.
http://www.nch.com.au/talk/index.html
The command-line version of Linphone may also be worth a look; the Windows GUI version is not accessible.
Please let me know if you find anything better.
Chris
On Tue, Sep 19, 2017 at 04:30:22PM -0400, Katherine M. Moss wrote: > Hello all, > > I forget what the concensus was when we discussed this before, > but I seem to be finding difficulties finding a softphone that > has both accessibility as well as a good feature set; > particularly, I need the ability to pull contacts and what not > from Active Directory since we use that; I've heard of Microsip, > though I know that it does not support this feature. If > X-Lite/Bria used to be accessible, they aren't now; unlabeled > fields everywhere which cannot be seemingly deciphered via > object navigation with NVDA (Haven't tried JAWS, but I will.) We're
using Asterisk for the server if that matters. Thanks.
> > _______________________________________________ > Blind-sysadmins mailing list > Blind-sysadmins@lists.hodgsonfamily.org > https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
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