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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