[Freeswitch-users] One-way audio when connecting incoming SIP call and WebRTC call using Valet Parking

[Gonzalo Gasca Meza]

Hi,
That's not a very simple call flow, and definetly more information is
required:

a) Browser version
b) FS version
c) Based on your description I guess FS1 is doing the transcoding from G711
<-> Opus?
d) No audio from Phone, do you see FS2 and FS1 sending RTP packets to
Browser?
e) Can you see incoming RTP packets at Browser from FS1?

"It result Handphone user can hear Browser user, but browser user cannot
hear handphone user."
I would start hooking up a phone in FS2, then FS1 and see if the same issue
occurs.

-G



On Wed, Feb 15, 2017 at 8:35 PM, Donny Hardyanto <hardyanto.donny at gmail.com>
wrote:

> Hi,
>
> I have problem got one-way audio when connection incoming SIP call and
> WebRTC call using ValetParking.
> This is my setup:
>
> Browser(WebRTC/Opus)->FS Box 1->FS Box 2 (Core)<-SBC(SIP/G711)<-Telco
> Network<-Handphone.
>
> The call came from SBC as SIP call and FS Box 2 accept and park call on
> valet parking.
> Browser place call to FS Box 1 and FS Box 1 will bridge to FS Box 2, and
> in FS Box 2, this call will retrieve call from valet parking.
>
> It result Handphone user can hear Browser user, but browser user cannot
> hear handphone user.
>
> How to solve this problem?
>
> Thanks,
>
> Donny
>
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