[SIP Call] inbound voice 3-4sec delay

ybenelli

ybenelli

New Member
Hi, we have AWS WCS 5 small and we having issues with sip calls,

there is a 3 or 4 seconds delay on inbound voice, there is not problem with outbound voice
The quality is good, the only problem is the incoming voice delay.

The WCS and PBX are on same region, the problem is the same on incomming call and on outcomming call, no matters if we call between internals or internal-external calls.

We test all codec and configuration on this post

Incoming voice issue on webrtc sip call

Hello, We have an issue while using demo interface for testing. In the phone interface we entered our sip details and made call. The voice of mine is going well to called user, however the voice of tha user is not well in our side. The voice is not understandable, not clear and dashed. Is...
forum.flashphoner.com

and we can't resolve it
sorry if my english is bad, thanks
 
Leonardo

Leonardo

New Member
Hello, I am working with Yonathan.
On the demo server we found the same issue. Is this a safe channel for us to share you the connection information to SIP Server?
I could share pcap file with the call.
Thaks, regards.
 
Max

Max

Administrator
Staff member
You can send private information via this form

1. pcap
2. Two SIP credentials for a call through your PBX. We should be able to pass success Softphone-Softphone call, like Linphone - Linphone through your PBX.
 
Leonardo

Leonardo

New Member
Thanks Max, I have submitted the info requested.
I will await for further indications, kinds regards.
 
Max

Max

Administrator
Staff member
We confirm a 1-1.5 seconds delay in browser, raised the ticket WCS-4033, will let you know about progress.
Please note: you do not provide SSH access to the server, so we can't use your server settings. We use the following parameters to reproduce the problem:
Code: 
codecs_exclude_sip=mpeg4-generic,flv,mpv,h265,opus,g729,speex16,g722
sip_force_tcp=true
allow_outside_codecs=false
profiles=42e01f,640028
because seems like your PBX does not support Opus codec.
Please also note that outgoing calls from 107 to 101 don't work for us even via softphones like Linphone or Bria, but 101 to 107 calls are establishing correctly.
 
Leonardo

Leonardo

New Member
Thanks Max, we wait for the update. Will now check the issue with the calls you experienced.
I will also send you the access to our flashphone server.
 
Leonardo

Leonardo

New Member
I have send you the acces, and open ssh port 22.
We tried calls between 107 and 101 had no issues.
We use xlite on 101 and microsip on 107, udp.
 
Max

Max

Administrator
Staff member
Seems like the problem is not in server settings, they are just slightly different. We will investigate it and let you know about progress.
 
Leonardo

Leonardo

New Member
Thanks for clarification. Is there any other kind of support for issues or incidents of production enviroments that requieres sla with our customers? Thanks un advance
 
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