Question on Web Call Server and Asterisk

Discussion in 'Web Call Server 3' started by roman, Dec 4, 2014.

  1. roman

    roman New Member

    Trying to configure Flashphoner for calls via Asterisk using HTML5 websockets.
    I open PhoneJS.html in Firefox and try to call, but get this error instead:
    Code:
    chan_sip.c process_sdp: Received AVP profile in audio offer but AVPF is enabled: audio 31002 RTP/AVP 8 0 18 100 111
    chan_sip.c:process_sdp: Failing due to no acceptable offer found
    Last edited: Dec 4, 2014
  2. Max

    Max Administrator Staff Member

    WCS does not require WebRTC support from the Asterisk-server, not does it requires any specific setup of AVPF. Web Call Server will interact with Asterisk vi standard SIP and RTP protocols and when working with a browser it uses WebRTC(SRTP/AVPF). Therefore you don't need to configure in Asterisk aside from common SIP peer configuration of course.
  3. roman

    roman New Member

    Do you have an exmaple of a wortking SIP-HTML5 phone?
  4. Max

    Max Administrator Staff Member

  5. I am trying a webrtc-sip via Asterisk call with Asterisk 14 and WCS Server version FlashphonerWebCallServer-5.0.2159. I am using the demo application for WebRTC(Phone and Phone Video) and Bria for SIP End Point. I am getting the following issue in the console of Asterisk
    [Apr 5 15:36:51] ERROR[61043][C-00000001]: chan_sip.c:5933 dialog_initialize_dtls_srtp: No DTLS-SRTP support present on engine for RTP instance '0x7f3614009500', was it compiled with support for it?
    [Apr 5 15:36:51] NOTICE[61043][C-00000001]: chan_sip.c:26201 handle_request_invite: Failed to authenticate device "1060" <sip:1060@192.168.30.156>;tag=fce5ad16
    Can anyone please help me on this since I am unable to debug and the same and I'm stuck with this since a couple of days.

    This is my http.conf
    [general]
    enabled=yes
    bindaddr=0.0.0.0 ; Replace this with your IP address
    bindport=8088 ; Replace this with the port you want to listen on
    ;sip.conf
    [general]
    realm=192.168.30.156 ; Replace this with your IP address
    udpbindaddr=192.168.30.156 ; Replace this with your IP address
    transport=udp

    [1060] ; This will be WebRTC client
    type=friend
    username=1060 ; The Auth user for SIP.js
    host=dynamic ; Allows any host to register
    secret=password ; The SIP Password for SIP.js
    encryption=yes ; Tell Asterisk to use encryption for this peer
    avpf=yes ; Tell Asterisk to use AVPF for this peer
    icesupport=yes ; Tell Asterisk to use ICE for this peer
    context=default ; Tell Asterisk which context to use when this peer is dialing
    directmedia=no ; Asterisk will relay media for this peer
    nat=no
    canreinvite=no
    qualify=no
    transport=udp,ws ; Asterisk will allow this peer to register on UDP or WebSockets
    force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11
    dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
    dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
    dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS cert file is
    dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS private key is
    dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when setting up DTLS
    dtlscafile=/etc/asterisk/keys/ca.crt
    allow=alaw
    media_encryption=dtls

    [1061] ; This will be the legacy SIP client
    type=friend
    username=1061
    host=dynamic
    secret=password
    context=default
    disallow=all
    allow=alaw
    qualify=no
    ;rtp.conf
    [general]
    rtpstart=10000
    rtpend=20000
    icesupport=yes
    stunaddr=stun.l.google.com:19302
    ;extensions.conf
    ;extensions.conf
    [default]
    exten => 1060,1,Dial(SIP/1060) ; Dialing 1060 will call the SIP client registered to 1060
    exten => 1061,1,Dial(SIP/1061) ; Dialing 1061 will call the SIP client registered to 1061
    Have checked the configs in Asterisk and it seems to be fine.
    Also I just wanted to know if the FlashPhoner client can support SRTP since I don't see the Offer/Answer having the fingerprint.
    Getting the following error in the Asterisk debug Logs
    han_sip.c:10726 process_sdp: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio
    Attached the logs for the WEB-SIP and SIP-WEB logs.
    Can anyone give an insight on this?Thanks for the help in advance.Really appreciate the help.

    Thanks,
    Karthik

    Attached Files:

    Last edited: Apr 6, 2017
  6. Max

    Max Administrator Staff Member

    Hello
    Web Call Server communicates with Asterisk as a legacy SIP peer and operates with browser as WebRTC peer.
    Browser <-- WebRTC --> WCS <-- legacy SIP/RTP --> Asterisk PBX
    So you have to configure peers for WCS as legacy SIP peers on your Asterisk server.

    Please create a new topic in the Web Call Server 5 branch of forum. This branch is out of date.

Share This Page