SIP video call from WCS can't hear audio of audio file was play from SIP server

manh.chu

Member
Hi admin,

I have a call script that is called internal when a sip peer (8001) calls to extensions (999999) it goes to IVR, here will play audio. Then goes to queue (2000), the queue has music on hold (moh). Next connects to sip peer (8000).

I tested SIP audio call and the video call does not pass WCS it works. I tested a manual audio call from WCS it worked, can hear the audio of IVR and moh of the queue and connect to peer 8000.

I have a problem when I make a video call from WCS it doesn't hear audio of IVR also moh of the queue. I capture pcap at WCS server and I see that SIP server sends RTP packet to WCS server. I extracted from the pcap of this call, it has audio of IVR and moh of the queue.

How can I fix it?

Bellow it is flow traffic when I capture of video call from WCS, in this the call not yet connect to peers 8000, RTP packet was sent from sip is the audio of IVR and moh of the queue.

1684593786470.png
 

Max

Administrator
Staff member
Hello

We raised internal ticket WCS-3825
It seems this issue is the same as described here
We will inform once it is fixed.
 

manh.chu

Member
Hello

We raised internal ticket WCS-3825
It seems this issue is the same as described here
We will inform once it is fixed.
Yes.
Thank you for reply.
 

Max

Administrator
Staff member
Good day.
The issue was fixed in build 5.2.1672. Please update WCS, then enable a/v generator
Code:
generate_av_for_ua=all
 

manh.chu

Member
I have a question about recording a SIP video call. WCS in the future have a plan for this feature.
Because of recording SIP video calls, I only recorded the direction of the caller.
 

Max

Administrator
Staff member
I have a question about recording a SIP video call. WCS in the future have a plan for this feature.
Because of recording SIP video calls, I only recorded the direction of the caller.
SIP calls recording is supported only for SIP as stream or SIP as RTMP functions. If you're using WebRTC-SIP gateway, the only way to record is to setup call recording at SIP PBX side.
There is also the ticket WCS-3723 to fix SIP calls audio recording, but no ETA for this ticket yet due to more priority tasks.
 

manh.chu

Member
Hi Max,

Today I test SIP video call again with config generate_av_for_ua=all . It has a problem with remote video that doesn't show, it has black. I capture pcap and see H264 packet is sent. I try to disable the config generator, remote video is shown.
I using WCS latest version: 5.2.1677.

1686797363004.png
 
Last edited:

Max

Administrator
Staff member
We reproduced the issue and working on it in the ticket WCS-3837. Let you know about progress here.
A quick and dirty workaround is to disable A/V generator, but in this case only video to video and audio to audio calls will work.
 

Max

Administrator
Staff member
Good day.
The issue was fixed in build 5.2.1691, now video calls should work correctly with generate_av_for_ua=all. Please update and check.
 

manh.chu

Member
Good day.
The issue was fixed in build 5.2.1691, now video calls should work correctly with generate_av_for_ua=all. Please update and check.
Yes, I test again, with generate_av_for_ua=all sip video call manual it works, with SIP video call with the script above I can hear the audio file but the remote video is still black, I capture flow see that rpt video packet transferred to WCS.
1688953856136.png
 

Max

Administrator
Staff member
We test a call from Phone Video example to Phone Video in Chrome desktop browser, and SIP as RTMP call (with both audio and video enabled) to Phone Video example in Chrome desktop browser. We can't reproduce the issue with build 5.2.1691.
with SIP video call with the script above
Please clarify what script do you use and what do you exactly doing, step by step.
 

manh.chu

Member
We test a call from Phone Video example to Phone Video in Chrome desktop browser, and SIP as RTMP call (with both audio and video enabled) to Phone Video example in Chrome desktop browser. We can't reproduce the issue with build 5.2.1691.

Please clarify what script do you use and what do you exactly doing, step by step.
the script I using is: I test an internal call when a sip peer (8001) calls to extensions (999999) it goes to IVR, and here will play audio. Then goes to queue (2000), the queue has music on hold (moh). Next connects to sip peer (8000).
 

Max

Administrator
Staff member
Please provide for us to test and reproduce the issue:
- two SIP accounts (caller and callee)
- the detailed description of the operation flow using two Phone Video examples
Please use this form to send.
 

Max

Administrator
Staff member
We reproduced the issue on our test server. Raised the ticket WCS-3866 to investigate and fix it. Please do not shut down the accounts, we'll use them for reproducing and testing.
 

manh.chu

Member
We reproduced the issue on our test server. Raised the ticket WCS-3866 to investigate and fix it. Please do not shut down the accounts, we'll use them for reproducing and testing.
Can you give me progress for this ticket? Thanks you.
 
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