[Freeswitch-users] Audio Connection Delay with SCA

[Sean Devoy]

FYI: New System w/ 2 Hyperthreaded Dual core 3.0Ghz XEON processors (vs P4
3Ghx) appears to have solved this delay problem.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Friday, June 08, 2012 4:00 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Audio Connection Delay with SCA

from FS point of view there should be little different from SCA and not SCA
besides the presence signaling, you should try some on-site testing with the
same phones.


On Fri, Jun 8, 2012 at 2:52 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
> Anthony,
> Thanks for the reply.
>
> We use all CSICO SPA504G phones.
> I did have the jitter buffer set up "HIGH" (choices: Low, Med, High, 
> Very High, Extremely High).  I dropped them down to Low now for 
> testing.  They do have pretty good network speeds (business level cable
modem).
>
> There is also something labeled:  " Jitter Buffer Adjustment:  "  
> Choices (Up, Down, Both, Disabled).  I have both.
>
> However, the issue is present, but far less pronounced in non-shared
lines.
> They have always said there was a connection delay at this site, but 
> with SCA it seems to be compounded.
>
> Maybe Group Call Pick will work better for them, they are all in the area.
>
> Sean
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of 
> Anthony Minessale
> Sent: Friday, June 08, 2012 3:28 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Audio Connection Delay with SCA
>
> what type of phone?  If it has a jitter buffer, you can try reducing 
> or disabling it.
> its probably a combo of jitter buffers charging up on the phone, and 
> the auto-ajust trying to fix the broken sdp for NAT.
>
>
> On Fri, Jun 8, 2012 at 12:12 PM, Sean Devoy <sdevoy at bizfocused.com> wrote:
>> Hi,
>>
>>
>>
>> I finally got Shared Call Appearance working to 4 NAT connected
> phones!!!!!
>>
>>
>>
>> Now the users say when they answer that line, they must wait a second 
>> or so to start talking (no big deal), but they all claim it takes 
>> nearly 5 seconds to be able to hear the caller.  They answer the 
>> standard “Thank You for calling XYZ, how can I help you today” which 
>> the caller DOES HEAR, but they cannot hear his response for the first
>> 5 seconds and ALWAYS have to ask them to repeat it.
>>
>>
>>
>> In the logs, I see what I believe is them picking up extension 299:
>>
>>     2012-06-08 11:44:15.115249 [NOTICE] switch_channel.c:926 New 
>> Channel
>> sofia/external/sip:299 at 69.251.170.6:5063 

>>
>> And then the RPT finally connecting:
>>
>>    2012-06-08 11:44:20.665094 [DEBUG] sofia_glue.c:3248 AUDIO RTP 
>> [sofia/external/sip:299 at 69.251.170.6:5064] 10.10.40.185 port 23500 ->
>> 69.251.170.6 port 16498 codec: 0 ms: 20
>>
>>
>>
>> Note the 5.5 second delay!  Granted there are 100+ lines between 
>> those if you want to see them, I will post them.
>>
>> After that audio is fine.
>>
>>
>>
>> Any thoughts would be appreciated.
>>
>> Sean
>>
>>
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>>
>
>
>
> --
> Anthony Minessale II
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>  
> 
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>
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>  
> 
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--
Anthony Minessale II

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