[asterisk-users] Asterisk 13.2.0 Video issues

Toufic Khreish (Gmail) toufic.khreish at gmail.com
Wed Mar 11 09:45:23 CDT 2015


Should I unload or rename the res_format_attr_h264.so        H.264 Format
Attribute Module
The asterisk server 13.2.0 does not break anymore upon calls towards GXV3175
grandstream, however only downstream video displayed on the GXV3175 is very
slow (1 frame per 10 seconds)

This problem only concerns GXV3175 for the moment (with the
res_format_attr_h264.so removed). (GXV3175 version  Hardware : 1.4A ,
program version: 1.0.3.76 and CPE version 1.0.1.32)

Any idea why ? and how could this be fixed ?


-----Original Message-----
From: Toufic Khreish (Gmail) [mailto:toufic.khreish at gmail.com] 
Sent: Tuesday, March 10, 2015 11:23 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Asterisk 13.2.0 Video issues

Thank you, I needed a starting point to start my post.

1. Asterisk 12.8.1 (IAX2 voice issues) no video issues.
Voice issues on IAX2 Trunks, All extensions are SIP.
The IAX2 trunks on Asterisk 12.8.1 produces only  one error out of : iax2
set debug trunk on
[2015-03-10 16:28:42] WARNING[9614][C-0000000b]: chan_iax2.c:1793
compress_subclass: Can't compress subclass 2097217

On the box running asterisk 1.6.2.6 I receive the following warning:
[2015-03-10 16:35:00] WARNING[24872]: translate.c:168 framein: no samples
for alawtolin


core show channels
Channel              Location             State   Application(Data)
IAX2/Mypbx1-15288    (None)               Up      AppDial((Outgoing Line))
SIP/6000-0000000f    (None)               Up
Dial(IAX2/Mypbx1/300,300,Tt)
2 active channels

Trunks are between an asterisk 1.6.2.6 and asterisk 12.8.1 (IAX , Alaw and
GSM codecs) Voice is not very clear and choppy 

If I try the same between an asterisk 13.2.0 and the asterisk 1.6.2.6 ,
voice is very clear.

2. Asterisk 13.2.0 Video issues (no IAX2 voice issues).

Calls from Bria video sip phone (android or IOS) to Grandstream GXV3175
(asterisk engine stops/crashes) Call from Groundwire video sip (IOS since
Android version does not H264 codec) to Grandstream GXV3175, Asterisk stops

Calls between SIP Video softphones works well no issues.
Calls from SIP video softphones (BRIA) to Grandstream GXV3275 works well.
(Acrobitz Groundwire to GXV3275 picture on Grandstream is yellowish) Calls
between GXV3275 and GXV3175 video streaming is very slow on the GXV3175
(this is not the case under Asterisk 12.8.1) Calls from GXV3175 to Bria
(video is displayed on bria side only)

There might be an issue on the Grandstream sip video phones as far as H264
is concerned however the case of streaming slowness is not there under
Asterisk 12.8.1) I cannot find anything related to the moment where asterisk
is breaking upon calling GXV3175

Best regards
Khreish Toufic


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Matthew Jordan
Sent: Tuesday, March 10, 2015 1:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.2.0 Video issues

On Tue, Mar 10, 2015 at 4:15 AM, Toufic Khreish (Gmail)
<toufic.khreish at gmail.com> wrote:
> I recently compiled asterisk 13.2.0 on an RK3288 , I am suspecting 
> problems with the format H264, Asterisk 12.8.1 compiled on the same 
> hardware is behaving very well for the same format H264
>
> Problem of asterisk 12.8.1 is IAX2 trunk bad voice quality.
>
> Could someone investigate the problem of Asterisk 13 with video 
> support on
> H264 ?
>

There's no where near enough information in your e-mail to give someone an
indication on where to start.

What channels are involved? What are their configurations? What formats are
negotiated on the channels? What symptoms do you see? What does the CLI
show, both when active calls are running and for a 'core show channel' for
the involved parties?

--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:
http://digium.com & http://asterisk.org

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