Integration issue for WebRTC with WCS server 5 and Asterisk 14.

Discussion in 'Web Call Server 5' started by Aghanash Karthik, Apr 6, 2017.

  1. Max

    Max Administrator Staff Member

    Please share details: screenshots, errors, etc.
  2. Max

    Max Administrator Staff Member

    For the details related iOS samples building, please create a new forum thread.
  3. Hi Max,
    Thank you for the response. I just wanted to know if you could let me know as to where I can get the latest version of phonejs since we are developing the entire webrtc version on that. I can see that the video calls are not working. The devices for video is not getting shared.It seems that that the sdp has a=recvonly. I request you to please let me know if I could find the latest version. Thank you for your time and help in advance..:)
  4. Can you let me know where I can find the latest sdk for the below mentioned UI.
    upload_2017-4-19_0-58-59.png
  5. Max

    Max Administrator Staff Member

    Hello
    This UI is out of date and not currently supported. If you have a valid support contract for legacy versions (wcs3), please contact support@flashphoner.com with any questions.
    Please use the latest Web SDK: https://flashphoner.com/wcs-web-sdk
    It is supported through the support forums.
  6. Thank you for the response. I am aware that this sdk is outdated. I just wanted to know the reason for not working. I just wanted to know since the getUserMedia for video is failing and the browser is sharing only the audio devices but the video devices are not getting shared. I have downloaded the WCS client from the following link
    https://flashphoner.com/downloads/builds/flashphoner_client/wcs-3.0-video/WCS-client-3.0.519-f90da2db92dfefaf2cee29e2e9962cdd58825a03.tar.gz
    The browser gives a prompt but the video share does not happen and it fails. Can you please help me out on this?
  7. Max

    Max Administrator Staff Member

    Unfortunately we can't commit our engineers to the old SDK and investigate why it does not work. We need a strong reason to do that.
    If you are our existing customer and if you have a valid support contract, please contact support@flashphoner.com for further support.
    If no, please use our latest available SDK.
  8. Hi Max,
    Thank you for the response. I am trying the same setup on aws with the WCS server running on the aws instance with ubuntu 16.04 installed on it. Since it is Natted, it has 2 IPs one private and the other public. I am having an Asterisk PBX which is also hosted on AWS. This also has 2 IPs. One private and the other public IP. I can see that the registration is not happening from the demo App to the Asterisk. The ICMP reachability seems to be fine for all the instances. Any specific configuration changes that I need to amke on the WCS server for the registration and the signalling and media to happen successfully. I request you to help me on this. Thank you for your help in advance..:)
  9. Max

    Max Administrator Staff Member

    Hello

    You have to open the following ports:
    WCS
    UDP 30000-32000
    8443 TCP
    Asterisk
    ? UDP
    5060 UDP

    If you still have issues, please
    1. Check your Asterisk server registration with a desktop softphone like Xlite.
    2. Share the pcap dump:
    Code:
    tcpdump udp -s 4096 -w log.pcap
    As an option you can get preconfigured WCS instance from here:
    https://aws.amazon.com/marketplace/pp/B01D1L5EAK
    and update wcs on this preconfigured instance
    Code:
    service webcallserver update
    Recommended settings for integration with Asterisk on Amazon (flashphoner.properties):
    Code:
    rtc_ice_add_local_component=true
    client_mode=false
    Last edited: Apr 21, 2017
  10. I can see that the sip phone is sending the RTP but the webserver fails to send as seen from the channelstats in asterisk
    2 active SIP channels
    ip-172-31-24-111*CLI> sip show channelstats
    Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
    174.62.206.166 65cda1c11e0 00:01:10 0000002546 0000000403 (13.67%) 0.0000 0000000000 0000000000 ( 0.00%) 0.0073
    34.204.51.242 44323dc0-26 00:01:10 0000000000 0000000001 (100.00%) 0.0000 0000002671 0000000000 ( 0.00%) 0.0000
    2 active SIP channels
    ip-172-31-24-111*CLI> sip show channelstats
    Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter
    174.62.206.166 65cda1c11e0 00:01:10 0000002566 0000000403 (13.57%) 0.0000 0000000000 0000000000 ( 0.00%) 0.0073
    34.204.51.242 44323dc0-26 00:01:11 0000000000 0000000001 (100.00%) 0.0000 0000002691 0000000000 ( 0.00%) 0.0000

    Can you please help on this?Thanks for your help..:)
  11. HI Max,
    This is the configurations that is being done on aws webcall server in the inbound security groups
    upload_2017-4-21_5-37-29.png
    The outbound traffc is given by as follows
    upload_2017-4-21_5-39-8.png
    Can you please let mw know if I have missed out on anything..?Thanks for the help..:)
  12. Max

    Max Administrator Staff Member

    Please provide pcap from WCS server
    Code:
    tcpdump udp -s 4096 -w log.pcap

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