Help on the compatibility of FLASPHONER WCS3.1312 latest release.

Discussion in 'Web Call Server 3' started by Alain, Sep 3, 2015.

  1. Alain

    Alain New Member

    Firstly, I think Alex for his effort to resolve our issue after installing the the latest WSC3
    • Can anyone advice for the best SIP Server to use from all on the market ?
    • Is a real SIP server like http://www.brekeke.com/sip/ whom claims WebRTC compatible ? would be more suitable than standard Asterisk.
    • With WSC 3 on Chrome not sending voice (data) yet on IE11 is great sound like a Dolby Studio with Windows10
    • Now, when it's Vista / Win7 / 8 then Chorme functions perfectly but IE 10 & 11 out of order and even blocks NO hungup at all.
    • Which is better the Asterisk SIP or Free-Switch... How about Elastix they also claim compatibility with WebRTC...
    • Would the use of PSTN routing line resolve the various issues for Flashphoner...
    Here are few lines around main error:
    a=rtcp:31694 IN IP4 37.59.28.96
    a=sendrecv
    a=ssrc:213118921 cname:rtp/audio/RTC-5052330277293314ad3913336a0815fd@37.59.28.96
    WebRtcMediaManager - onSetRemoteDescriptionErrorCallback(): error: Failed to set remote answer sdp: Called in wrong state: STATE_INPROGRESS
    UTC 16:59:38.646 - Click2Call - notify call_id: 5052330277293314ad3913336a0815fd@37.59.28.96 call.anotherSideUser: undefined
    UTC 16:59:38.654 - stopSound soundName: RING me: [object Object] me.ringSound: [object HTMLAudioElement]
    Can you guys give us your opinion and the server you are using to fully enjoy Flashphoner...
  2. Max

    Max Administrator Staff Member

    Hello,
    This error is related WCS3 only
    Code:
    a=rtcp:31694 IN IP4 37.59.28.96
    a=sendrecv
    a=ssrc:213118921 cname:rtp/audio/RTC-5052330277293314ad3913336a0815fd@37.59.28.96
    WebRtcMediaManager - onSetRemoteDescriptionErrorCallback(): error: Failed to set remote answer sdp: Called in wrong state: STATE_INPROGRESS
    UTC 16:59:38.646 - Click2Call - notify call_id: 5052330277293314ad3913336a0815fd@37.59.28.96 call.anotherSideUser: undefined
    UTC 16:59:38.654 - stopSound soundName: RING me: [object Object] me.ringSound: [object HTMLAudioElement] 
    We can't reproduce it with WCS4.
    Do you mean WCS4?
    With WCS4 you should uncomment line in the Click-to-Call-min.html page:
    Code:
    <script type="text/javascript" src="../../dependencies/swf/swfobject.js"></script>
    Then it should work with IE 10, 11 and any other browser supporting latest Flash Player.
  3. Max

    Max Administrator Staff Member

    Can anyone advice for the best SIP Server to use from all on the market ?
    There are a lot of SIP gateways and SIP proxy on the market.
    I would use a SIP server for that I have a skilled network engineer in my team.
    WCS4 is compatible with any SIP server and has a lot of configuration options to be tuned for compatibility.
    Compatibility list:
    • Asterisk
    • OpenSIPs
    • Communigate Pro
    • Porta Switch
    • Avaya
    • Cisco Call Manager
    • Free Switch
    etc
    If you are using WCS, you absolutely don't need WebRTC support in SIP server. WCS is designed to handle all WebRTC streams and calls. So it can be any SIP server supporting SIP, RTP/AVP legacy protocols.

    Code:
    With WSC 3 on Chrome not sending voice (data) yet on IE11 is great sound like a Dolby Studio with Windows10
    It is a bug with in Chrome or WCS3 WebRTC implementation. However, WCS3 is currently out of support. So you can try WCS4 where this issue is fixed.
    Please see my previous reply regarding WebRTC capabilities. If we are talking about WCS, you don't need to configure SIP server for WebRTC.
    WCS4 works properly. If you have any issues, let us know.
  4. Alain

    Alain New Member

    Thank you Max for all your replies on Chrome browser issue. Best Regards Alain

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