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@lists.freeswitch.org [mailto:freeswitch-users-bounces@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@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@lists.freeswitch.org [mailto:freeswitch-users-bounces@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@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@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@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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