Incoming voice issue on webrtc sip call

[phonedev]

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 there a configuration we missed on flashphoner.properties?

 

[Max]

Good day.
Seems like this is a codec or bandwidth issue. Please check what audio codec is set in softphone parameters of the other side. Try to set up Opus codec if your SIP PBX supports that.
You can also pass an additional parameters in INVITE SDP to manage bandwidth. Please read details here.

 

[phonedev]

Hello Max,
Thank you for your suggestions. But we couldn't get success yet.
The SIP provider codecs are g711 alaw or g711 ulaw. They are not using opus codec.
Also I added additional parameters in INVITE SDP as in documentation, not solved our problem.
Let me share our flashphoner.parameters too if there are misconfiguration, maybe it helps:

#server ip
ip                     = #ourserverip
ip_local               = #ourserverip

#domain
domain                 = #ourdomain
#webrtc ports range
media_port_from        =31001
media_port_to          =32000

#codecs
codecs                   =opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv,g711
codecs_exclude_sip       =mpeg4-generic,flv,mpv
codecs_exclude_streaming =flv,telephone-event
codecs_exclude_sip_rtmp  =opus,g729,g722,mpeg4-generic,vp8,mpv,g711,alaw,ulaw

#websocket ports
ws.port                 =8080
wss.port                =8443

client_mode=false

rtc_ice_add_local_component=true

dtmf=INFO

 

[Max]

client_mode=false
rtc_ice_add_local_component=true

These two parameters say that you are on Amazon EC2 instance.

1. Make sure your instance is running in the same region as region of your SIP PBX.
2. Make sure your instance has enough resources. For example, t3.micro is not a proper server for tests.

You can also check call with our demo server https://demo.flashphoner.com

Phone Min

This server is running in Europe/Frankfurt region.

 

[phonedev]

Hello Max,

I removed that lines from properties. Our server instance is running in Europe/Amsterdam, we are using Vultr.
Instance has 1 vCore, 2GB ram and 55GB SSD. I guess these are matches minimum requirements in documentation. Do you have any otehr suggestion?

 

[Max]

I removed that lines from properties. Our server instance is running in Europe/Amsterdam, we are using Vultr.

The lines you've removed will be added again on server restart if your hosting provider exposes Amazon-like API because this is necessary settings if your instance is behind NAT.
Please make sure your SIP PBX is in the same region.
Please also check the call with our demo server as we recommended above. If the issue still persists please provide SSH access to your instance and two SIP accounts to test a call via your SIP PBX. Use this private form to send all the credentials.

 

[phonedev]

Hello Max,

I tried your demo server yesterday. Our sip totally not worked with your server. I shared our server information and sip info with your shared form.

 

[Max]

We tried to test SIP calls with credentials you've provided.
First, you SIP PBX does not support SIP TCP signaling. In this case, you should exclude unneeded codecs for SIP call to be established properly:

allow_outside_codecs=false
codecs_exclude_sip=mpeg4-generic,flv,mpv,h264,vp8,opus,speex16

Also, we need two SIP accounts to call from one to another (one from WCS and another from softphone for example). We cannot do that with SIP accounts you've provided: a call is not establihing.
So please provide two SIP accounts for test as we described, using this form.

 

[Max]

We've tested SIP calls with our WCS instance and with your WCS instance of course.
The problem you describe is not reproducing in our tests: audio quality looks good for both sides (calling from Phone Min example to Bria softphone), no delays, losses or dashes. Voice seems to be clear (and music too). Quality is slightly better with Opus codec (it seems to be supported by SIP PBX because we saw this codec in WebRTC internals chrome://webrtc-intenals).
So we recommend to exclude codecs you do not need and prevent browser from adding excessive codecs:

codecs_exclude_sip=mpeg4-generic,flv,mpv,h264,vp8,speex16,g729,g722
allow_outside_codecs=false

Also, remove g711 from codecs list because G711.alaw and G711.ulaw have their own keywords

With the settings above the following audio codecs will be used for SIP calls:
PCMA (G711.alaw)
PCMU (G711.ulaw)
OPUS
Also we recommend to check if the channel is good enough on callee side. For example, use 4G instead of 3G/2G, or Wi-Fi instead of 4G network. Check callee hardware (microphone).

 

[phonedev]

Thank you for your support Max, now both side voices are crystal clear.