Not a same quality of sound on both side..Why?

Pol Isidor

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Mar 12, 2019, 8:32:46 AM3/12/19

to dongle

Hello..

I'm using now ~one year asterisk v13.2x with dongle Huawei E1550 but I tried even other.

I'm using codec ULAW (tried even with g729 and ALAW).

The SIP client on mobile phone is CSIPSimple latest version. Whether I call someone ir I got the call, the other party do not hear me so nice and clean as I hear him. Often the other party asking to repeat the words, couse they said that it is unrecognizable. I tried with other mobile provider, with other dongle, with other mobile phone, with different internet (optic with low lattency), and even with different sip client.

What can cause this?

Let me be clear. The sound is not terrible but it is not so good as I hear other party.

Any suggestion?

Thx

The sip config is:

externhost=my_dns:_udp_port
localnet = 192.168.1.0/255.255.255.0
bindaddrr=0.0.0.0
bindport = udp_port
tcpbindaddr=0.0.0.0:tcp_port
tcpenable = yes
externtcpport = tcp_port
transport=udp,tcp
keepalive=yes
directmedia=no
defaultexpiry = 3600

canreinvite = no
subscribe_network_change_event = yes

disallow = all
allow = ulaw
nat=force_rport,comedia

textsupport=yes
accept_outofcall_message = yes                                                                                                                                  

[user]
context=incoming_CALL
outofcall_message_context=incoming_CALL
context=outgoing_CALL

host = dynamic
type = friend
mailbox=xxx@default
username=user
secret=xxxxx
qualifyfreq=20; How often to check is it peer online, same as traditional PING command!
qualify=5000; After how many milisecond from last ping to consider that peer is offline! 5000 send 10 rettransmision packets what is 5s, 2packet/sec
busylevel=1

and the dongle.conf is:

[general]

interval=5            ; Number of seconds between trying to connect to devices

;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
jbenable = yes            ; Enables the use of a jitterbuffer on the receiving side of a
                ; Dongle channel. Defaults to "no". An enabled jitterbuffer will
                ; be used only if the sending side can create and the receiving
                ; side can not accept jitter. The Dongle channel can't accept jitter,
                ; thus an enabled jitterbuffer on the receive Dongle side will always
                ; be used if the sending side can create jitter.

jbforce = no            ; Forces the use of a jitterbuffer on the receive side of a Dongle
                ; channel. Defaults to "no".

jbmaxsize = 200        ; Max length of the jitterbuffer in milliseconds.

jbresyncthreshold = 1000    ; Jump in the frame timestamps over which the jitterbuffer is
                ; resynchronized. Useful to improve the quality of the voice, with
                ; big jumps in/broken timestamps, usually sent from exotic devices
                ; and programs. Defaults to 1000.

jbimpl = fixed            ; Jitterbuffer implementation, used on the receiving side of a Dongle
                ; channel. Two implementations are currently available - "fixed"
                ; (with size always equals to jbmaxsize) and "adaptive" (with
                ; variable size, actually the new jb of IAX2). Defaults to fixed.

;jbtargetextra = 40        ; This option only affects the jb when 'jbimpl = adaptive' is set.
                ; The option represents the number of milliseconds by which the new jitter buffer
                ; will pad its size. the default is 40, so without modification, the new
                ; jitter buffer will set its size to the jitter value plus 40 milliseconds.
                ; increasing this value may help if your network normally has low jitter,
                ; but occasionally has spikes.

;jblog = no            ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------

[defaults]
; now you can set here any not required device settings as template
;   sure you can overwrite in any [device] section this default values

group=0                ; calling group
rxgain=0            ; increase the incoming volume; may be negative
txgain=0            ; increase the outgoint volume; may be negative
autodeletesms=yes        ; auto delete incoming sms
resetdongle=yes            ; reset dongle during initialization with ATZ command
u2diag=-1            ; set ^U2DIAG parameter on device (0 = disable everything except modem function) ; -1 not use ^U2DIAG command
usecallingpres=yes        ; use the caller ID presentation or not
callingpres=allowed_passed_screen ; set caller ID presentation        by default use default network settings
disablesms=no            ; disable of SMS reading from device when received
                ;  chan_dongle has currently a bug with SMS reception. When a SMS gets in during a
                ;  call chan_dongle might crash. Enable this option to disable sms reception.
                ;  default = no

language=en            ; set channel default language
smsaspdu=yes            ; if 'yes' send SMS in PDU mode, feature implementation incomplete and we strongly recommend say 'yes'
mindtmfgap=45            ; minimal interval from end of previews DTMF from begining of next in ms
mindtmfduration=8000        ; minimal DTMF tone duration in ms
mindtmfinterval=200        ; minimal interval between ends of DTMF of same digits in ms

callwaiting=auto        ; if 'yes' allow incoming calls waiting; by default use network settings
                ; if 'no' waiting calls just ignored
disable=no            ; OBSOLETED by initstate: if 'yes' no load this device and just ignore this section

initstate=start            ; specified initial state of device, must be one of 'stop' 'start' 'remote'
                ;   'remove' same as 'disable=yes'

dtmf=off

[dongle0]
audio=/dev/DONGLE-3G-MODEM-1        ; tty port for audio connection
data=/dev/DONGLE-3G-MODEM-2            ; tty port for AT commands
context=incoming_CALL              ; context for incoming calls

Marco Gaiarin

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Mar 12, 2019, 9:00:08 AM3/12/19

to chan_...@googlegroups.com

Mandi! Pol Isidor
In chel di` si favelave...

> Any suggestion?

Have you tried to play with:

> and the dongle.conf is:
[...]

> rxgain=0 ; increase the incoming volume; may be negative
> txgain=0 ; increase the outgoint volume; may be negative

?!

--
Marco ``Gaio'' Gaiarin | LUG Pordenone (http://pordenone.linux.it)
P.zza S. Tommaso, 20 | Lilliput BBS (http://bbs.lilliput.linux.it)
Cimpello di Fiume Veneto | Azione Cattolica - Concordia-Pordenone
33080 Pordenone (Italia) | (http://www.accanto.org)
Tel. +39-0434-56-1305 | http://www.gaiarin.it/ ga...@linux.it

Pol Isidor

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Mar 12, 2019, 1:06:05 PM3/12/19

to dongle

Maybe you didn't read enought clearly and with attention my post.

I didn't speak about volume level, that it is too high or to quite.

I spoke about quality, that it happening the degradation in sound on another party.

Marco Gaiarin

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Mar 13, 2019, 9:00:08 AM3/13/19

to chan_...@googlegroups.com

Mandi! Pol Isidor
In chel di` si favelave...

> Maybe you didn't read enought clearly and with attention my post.

Ahem, evidently yes. Sorry.

> I'm using now ~one year asterisk v13.2x with dongle Huawei E1550 but I
> tried even other.
> I'm using codec ULAW (tried even with g729 and ALAW).
> The SIP client on mobile phone is CSIPSimple latest version. Whether I call
> someone ir I got the call, the other party do not hear me so nice and clean
> as I hear him. Often the other party asking to repeat the words, couse they
> said that it is unrecognizable. I tried with other mobile provider, with
> other dongle, with other mobile phone, with different internet (optic with
> low lattency), and even with different sip client.
> What can cause this?
> Let me be clear. The sound is not terrible but it is not so good as I hear
> other party.
> Any suggestion?

In any case, before playing with asterisk jutterbuffers, tune the gain
using a SIP phone in LAN, because a bad-tuned audio can get things
worst...

This effectively seems a 'QOS' problem. Try to:

a) if possible, tune your 'router' QoS parameters.

b) try to enable (in sip.conf and/or in dongle.conf) the jitter buffer,
try:

jbenable = yes
jbforce = yes
jbmaxsize = 200
jbresyncthreshold = 1000
jbimpl = adaptive
; or try also:
; jbimpl = fixed

200ms is normally the theresold of 'echo'; if forcing the jitterbuffer
lead to better calls but you feel echo, try to lower 'jbmaxsize' to
reach a compromise.

Pol Isidor

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Mar 16, 2019, 2:48:34 PM3/16/19

to dongle

Yes mate..

Three days ago I played with this parameters and it seams that same as uyou suggest me is the right solution!

In any case thank you very much on idea.

What I enbled or changed in comparation with first dongle.conf are:

jbforce = yes
jbimpl = adaptive

The next parameter was active before too:

jbenable = yes
jbmaxsize = 200
jbresyncthreshold = 1000

It is now really big difference how the other party hear me, couse they stopped with: "Can you please repeat I didn;t understand the words" :)