Solved sip trunk to mobiltel / voice activity detection problem

Discussion in '3CX Phone System - General' started by Vesko, Sep 28, 2017.

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  1. Vesko

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    Dear all,

    just tested a 3cx 15.5 runing on Windows 2008 R2 server , with 1 sip trunk to voip provider Mobiltel in Bulgaria. Connectio to provider is done via separate LAN card via vlan from provider. Authentication is IP based from voip provider.

    Еverything is fine exept when we make OUTGOING call AND remote party doesn't have activity on microphone. In this scenario i hear clearly that voice activity detection supress a noice. Thise lead to inconvenience - after long pause first letters of remote party word are missed. I try to disable on phones VAD ,but it doesn't help. I try also with software phone from 3CX - still the same issue.

    All phones are in sepate vlan with qos, there is no losses

    All phones has Alaw codec which is the same for SIp trunk Alaw (no transcoding)

    When I got a INCOMING call , it can be heared that there is no voice activity detection and calls are perfect.

    I've got from provider information following information:

    voip signaling should be send to 213.226.61.66
    and
    voip media (rtp) should be send to 213.226.61.67

    I've separate LAN card with private IP from voip provider 10.253.2.XX. This has been added to SIP configuration of trunk for IP which will be used, and also added in localsubnets (to avoid nat)

    At the moment I think that voip media 213.226.61.67 should be used somewhere in SIp trunk, or I need to disable this voice activity detection somewhere.

    What can be the issue ?
     

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  2. sip.bg

    sip.bg Active Member

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    I've sent you a private message. Please call me.
     
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    #2 sip.bg, Sep 28, 2017
    Last edited: Sep 28, 2017
  3. sip.bg

    sip.bg Active Member

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    The issue is related to SIP trunk of the provider (Mobiltel, part of A1 group) and not to 3CX.
    VAD (Voice Activity Detection) suppression of voice stream is felt after periods of silence as minimal infringement of next piece of voice.
     
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    YiannisH_3CX likes this.
  4. Vesko

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    Thanks a lot for professional help from SIP.BG and 3CX community! He helped me to discover where is the problem. In our case - issue was with the provider of SIP trunk. 3CX is great PBX. Wireshark before PBX discover RTP (CN) which SIP trunk provider inject in stream during VAD...
     
    YiannisH_3CX and sip.bg like this.
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