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  • 1.  Twilio and WebRTC

    Posted 05-13-2020 06:38
    No replies, thread closed.
    This could be a stupid question as I know it is not officially supported.

    I am currently using Twilio as telco provider with virtual edge (it is an development / testing environment). I want to perform testing with phone. However, WebRTC supports only opus, but Twilio supports only g.711. What other solution(s) I can have? Actually using WebRTC is perfect since some developers / testers are using Mac.

    p.s. For softphone, we still have no luck to provision one. Our edge is current connected to Internet directly with no firewall in front. It can handle inbound calls but don't know why we can't provision softphone properly.
    #SIP/VolP
    #Telephony

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    Wong Ricky
    Individual Only Contact Account
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  • 2.  RE: Twilio and WebRTC

    Posted 05-13-2020 07:32
    No replies, thread closed.
    The Phone Trunks would be set up with Opus and the External Trunks with G.711. The Edge does the transcoding.

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    George Ganahl GCP (Genesys Cloud), ICCE
    Principal Technology Consultant
    Genesys
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  • 3.  RE: Twilio and WebRTC

    Posted 05-13-2020 10:35
    No replies, thread closed.
    Thanks. I've tried again and in-bound call works fine now !!

    The only challenge remained is out-bound call. Again, everything look fine, e.g. when I tried to simulate call in Site configure, I got "Success". However, when I tried to make outbound call, I always got busy tone. I have checked the logs from Edge, I found multiple INVITE requests but that's all - IP address is correct for both the source and destination. On Twilio's side, there is no logs about these outbound call - but I have double checked all access control settings should be setup properly. Please note that my Edge is not behind router nor firewall anymore and it connects to Internet directly, so there shouldn't be any firewall issue.

    Any suggestion on what I should check?

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    Wong Ricky
    Individual Only Contact Account
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  • 4.  RE: Twilio and WebRTC

    Posted 05-13-2020 10:49
    No replies, thread closed.
    Are you sure you are sending to the correct port and protocol for Twilio? What listen port is Twilio configured to use?

    Jim Crespinio posted a document for configuring Genesys Cloud to work with Twilio for BYOC:
    https://community.genesys.com/communities/community-home1/librarydocuments/viewdocument?DocumentKey=0592a918-5b05-4d7f-8bb3-d5eb3e72854c&CommunityKey=bab95e9c-6bbe-4a13-8ade-8ec0faf733d4&tab=librarydocuments

    It has info you can use to make sure you have both ends set up correctly (even though you are not using BYOC). Check the information linked on the Twilio site.

    (By the way...Twilio documentation says it supports TLS, so it sounds like you may need to look at the Twilio configuration more carefully...your other thread seemed to indicate Twilio "requires" UDP for SIP signalling, but that is not the case).

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    George Ganahl GCP (Genesys Cloud), ICCE
    Principal Technology Consultant
    Genesys
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  • 5.  RE: Twilio and WebRTC

    Posted 05-13-2020 11:08
    No replies, thread closed.
    Thanks !! Port number is the problem !! I have changed my in-bound call port number to another value, and PureCloud used that port number as default as well for outbound call, however, on Twilio side it expects the traffic will be come on standard port # (which is 5060). I set the port number properly in SIP proxy field and it works now !!

    Thanks a lot for kindly support.​

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    Wong Ricky
    Individual Only Contact Account
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  • 6.  RE: Twilio and WebRTC

    Posted 05-13-2020 11:13
    No replies, thread closed.
    I'm glad I could help

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    George Ganahl GCP (Genesys Cloud), ICCE
    Principal Technology Consultant
    Genesys
    ------------------------------