[Freeswitch-users] Bad sound with crackle

Jeffrey Leung curriegrad2004 at gmail.com
Sat Nov 13 00:01:04 PST 2010


Try changing your RTP packet time to 30ms and see what happens.

>  Hi Anthony,
>
>  I have this warning as well with my Siemens C470IP as my Linksys
>  SPA942.
>  And my Linksys has been configured with a RTP Packet Size to 0.020. If
>  I change to 0.20 I have the same warning :-/
>
>  Loïc
>
>
>
>  On Fri, 12 Nov 2010 11:23:46 -0600, Anthony Minessale
>  <anthony.minessale at gmail.com> wrote:
>> What is your device?
>>
>> is it a cisco/linksys/sipura ?
>>
>> you can change the packet size from 0.30 to 0.20 in the UI of your
>> phone.
>>
>>
>>
>>
>> On Fri, Nov 12, 2010 at 6:09 AM, Loïc <loic.latreille at ovh.net> wrote:
>>>  Hello,
>>>
>>>  I am a new user of FreeSwitch and I try to switch from Asterisk to
>>>  FreeSwitch.
>>>
>>>  For now, I just registered a SIP line and I made an extension in
>>> the
>>>  dialplan to playback music on hold when I call :
>>>
>>>  <context name="default">
>>>
>>>     <extension name="test">
>>>       <condition>
>>>         <!--<condition field="destination_number"
>>>  expression="^9000$">-->
>>>         <action application="answer"/>
>>>         <action application="playback" data="$${hold_music}"/>
>>>       </condition>
>>>     </extension>
>>>
>>>  </context>
>>>
>>>  When I call I have a problem with the sound, it is very bad, I hear
>>>  very fast crackle.
>>>  On the console I can see a warning during my call and that's all:
>>>
>>>  2010-11-12 13:02:45.119517 [NOTICE] switch_channel.c:784 New
>>> Channel
>>>  sofia/external/anonymous at anonymous.invalid
>>>  [c27034e2-ee54-11df-ad91-7fc19b304698]
>>>  2010-11-12 13:02:45.122548 [INFO] mod_dialplan_xml.c:331 Processing
>>>  Anonymous <anonymous>->test in context default
>>>  2010-11-12 13:02:45.125555 [NOTICE] mod_dptools.c:920 Channel
>>>  [sofia/external/anonymous at anonymous.invalid] has been answered
>>>  2010-11-12 13:02:45.508790 [WARNING] mod_sofia.c:1036 Asynchronous
>>>  PTIME not supported, changing our end from 30 to 20
>>>
>>>  If I add <X-PRE-PROCESS cmd="set" data="timer_name=soft"/> in
>>> vars.xml,
>>>  when I call the sound is good but I still get the warning :
>>>
>>>  2010-11-12 13:05:42.789161 [NOTICE] switch_channel.c:784 New
>>> Channel
>>>  sofia/external/anonymous at anonymous.invalid
>>>  [2c565760-ee55-11df-a0d9-0f91e325cd76]
>>>  2010-11-12 13:05:42.791179 [INFO] mod_dialplan_xml.c:331 Processing
>>>  Anonymous <anonymous>->test in context default
>>>  2010-11-12 13:05:42.794179 [NOTICE] mod_dptools.c:920 Channel
>>>  [sofia/external/anonymous at anonymous.invalid] has been answered
>>>  2010-11-12 13:05:43.174443 [WARNING] mod_sofia.c:1036 Asynchronous
>>>  PTIME not supported, changing our end from 30 to 20
>>>  2010-11-12 13:05:43.175461 [INFO] sofia.c:709
>>>  sofia/external/anonymous at anonymous.invalid Update Callee ID to
>>>  "anonymous" <anonymous>
>>>
>>>  Is this normal?
>>>  Why should I add this line in vars.xml for it to work ?
>>>  What does this warning? How to solve it?
>>>
>>>  Thank you in advance for your help.
>>>
>>>
>>>  Loïc
>>>
>>>
>>>
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>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
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