bug: sip to webrtc calling failed

[avoylenko]

Calls from locally a registered SIP phone fails with JsSIP Internal Error:

index.js:2 JsSIP:WARN:RTCSession emit "peerconnection:setlocaldescriptionfailed" [error:DOMException: Failed to execute 'setLocalDescription' on 'RTCPeerConnection': Failed to set local answer sdp: Failed to set local audio description recv parameters for m-section with mid='1'.]

SIP/2.0 500 JsSIP Internal Error
Via: SIP/2.0/WSS 54.236.168.131:8443;branch=z9hG4bKcv1g6Ngp8mc2c
To: <sip:gvuafv43@d056325m6dra.invalid>;transport=ws;tag=c8rr141gm7
From: <sip:1000@91.197.168.161:49368>;tag=Xa99X2SZXDpae
Call-ID: 1c9b947c-f856-123c-9c84-0e9252b57157
CSeq: 75142179 INVITE
Supported: timer,ice,replaces,outbound
Content-Length: 0

Steps to reproduce:

1. Start a SIP phone
2. Start a Jambonz Webphone phone
3. Place a call from the SIP phone to the Jambonz Webphone