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
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
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
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
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.
http://www.linphone.org/
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.
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