• Почему Webrtc Asterisk нет звука при входящих?

    @Enj0y Автор вопроса
    Дамп INVITE пару секунд, там сразу идёт приветствие, так что звук должен быть. К слову, ни трубки ни софт фоны с включенным DTLS входящие вообще не берут, возможно не поддерживают, либо дело в DTLS.
  • Почему Webrtc Asterisk нет звука при входящих?

    @Enj0y Автор вопроса
    Andrey Barbolin,
    2019/09/08 23:19:49.649826 127.0.0.1:8088 -> 127.0.0.1:59386
    INVITE sip:q2uv843c@pi278aeds572.invalid;transport=ws SIP/2.0
    Via: SIP/2.0/WS 127.0.0.1:5166;branch=z9hG4bK26864f4f;rport
    Max-Forwards: 70
    From: "79999999999" <sip:101@127.0.0.1:5166>;tag=as5bdd7a38
    To: <sip:q2uv843c@pi278aeds572.invalid;transport=ws>
    Contact: <sip:101@127.0.0.1:5166;transport=ws>
    Call-ID: 45c330626f23227e638200ef0fefc8c6@127.0.0.1:5166
    CSeq: 102 INVITE
    User-Agent: server
    Date: Sun, 08 Sep 2019 20:19:49 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    Call-Info: <sip:ip>;answer-after=0
    P-Asserted-Identity: "79999999999" <sip:101@127.0.0.1>
    Content-Type: application/sdp
    Content-Length: 697
    
    v=0
    o=root 356291339 356291339 IN IP4 127.0.0.1
    s=Asterisk PBX 16.5.1
    c=IN IP4 127.0.0.1
    t=0 0
    m=audio 14436 RTP/SAVPF 0 8 107 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:107 opus/48000/2
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=maxptime:60
    a=ice-ufrag:52c9d2671572a2395d96fed851dd58f2
    a=ice-pwd:62c9737c1f1c1af96da6020d19dd8075
    a=candidate:Hbca5df9e 1 UDP 2130706431 188.165.***.*** 14436 typ host
    a=candidate:Hbca5df9e 2 UDP 2130706430 188.165.***.*** 14437 typ host
    a=connection:new
    a=setup:actpass
    a=fingerprint:SHA-256 40:62:F0:F5:32:14:09:77:39:9C:08:66:24:E7:66:24:8D:DD:62:79:01:89:52:85:08:78:91:B9:36:9C:3C:39
    a=rtcp-mux
    a=sendrecv
    
    
    2019/09/08 23:19:49.712541 127.0.0.1:59386 -> 127.0.0.1:8088
    SIP/2.0 100 Trying
    Via: SIP/2.0/WS 127.0.0.1:5166;branch=z9hG4bK26864f4f;rport
    To: <sip:q2uv843c@pi278aeds572.invalid;transport=ws>
    From: "79999999999" <sip:101@127.0.0.1:5166>;tag=as5bdd7a38
    Call-ID: 45c330626f23227e638200ef0fefc8c6@127.0.0.1:5166
    CSeq: 102 INVITE
    Supported: outbound
    User-Agent: SIP.js/0.7.8
    Content-Length: 0
    
    
    
    2019/09/08 23:19:49.727988 127.0.0.1:59386 -> 127.0.0.1:8088
    SIP/2.0 180 Ringing
    Via: SIP/2.0/WS 127.0.0.1:5166;branch=z9hG4bK26864f4f;rport
    To: <sip:q2uv843c@pi278aeds572.invalid;transport=ws>;tag=omq79k93vm
    From: "79999999999" <sip:101@127.0.0.1:5166>;tag=as5bdd7a38
    Call-ID: 45c330626f23227e638200ef0fefc8c6@127.0.0.1:5166
    CSeq: 102 INVITE
    Contact: <sip:q2uv843c@pi278aeds572.invalid;transport=ws>
    Supported: outbound
    User-Agent: SIP.js/0.7.8
    Content-Length: 0
    
    
    
    2019/09/08 23:19:49.756947 127.0.0.1:8088 -> 127.0.0.1:59386
    ACK sip:q2uv843c@pi278aeds572.invalid;transport=ws SIP/2.0
    Via: SIP/2.0/WS 127.0.0.1:5166;branch=z9hG4bK1f58ea11;rport
    Max-Forwards: 70
    From: "79999999999" <sip:101@127.0.0.1:5166>;tag=as5bdd7a38
    To: <sip:q2uv843c@pi278aeds572.invalid;transport=ws>;tag=omq79k93vm
    Contact: <sip:101@127.0.0.1:5166;transport=ws>
    Call-ID: 45c330626f23227e638200ef0fefc8c6@127.0.0.1:5166
    CSeq: 102 ACK
    User-Agent: server
    Content-Length: 0
    
    
    
    2019/09/08 23:19:49.857263 127.0.0.1:8088 -> 127.0.0.1:59386
    INVITE sip:q2uv843c@pi278aeds572.invalid;transport=ws SIP/2.0
    Via: SIP/2.0/WS 127.0.0.1:5166;branch=z9hG4bK214a3abf;rport
    Max-Forwards: 70
    From: "79999999999" <sip:101@127.0.0.1:5166>;tag=as5bdd7a38
    To: <sip:q2uv843c@pi278aeds572.invalid;transport=ws>;tag=omq79k93vm
    Contact: <sip:101@127.0.0.1:5166;transport=ws>
    Call-ID: 45c330626f23227e638200ef0fefc8c6@127.0.0.1:5166
    CSeq: 103 INVITE
    User-Agent: server
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
    Supported: replaces, timer
    P-Asserted-Identity: "Александр" <sip:89999999999@127.0.0.1>
    Content-Type: application/sdp
    Content-Length: 697
    
    v=0
    o=root 356291339 356291340 IN IP4 127.0.0.1
    s=Asterisk PBX 16.5.1
    c=IN IP4 127.0.0.1
    t=0 0
    m=audio 14436 RTP/SAVPF 0 8 107 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:107 opus/48000/2
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=maxptime:60
    a=ice-ufrag:52c9d2671572a2395d96fed851dd58f2
    a=ice-pwd:62c9737c1f1c1af96da6020d19dd8075
    a=candidate:Hbca5df9e 1 UDP 2130706431 188.165.***.*** 14436 typ host
    a=candidate:Hbca5df9e 2 UDP 2130706430 188.165.***.*** 14437 typ host
    a=connection:new
    a=setup:passive
    a=fingerprint:SHA-256 40:62:F0:F5:32:14:09:77:39:9C:08:66:24:E7:66:24:8D:DD:62:79:01:89:52:85:08:78:91:B9:36:9C:3C:39
    a=rtcp-mux
    a=sendrecv
    
    
    2019/09/08 23:19:49.930674 127.0.0.1:59386 -> 127.0.0.1:8088
    SIP/2.0 100 Trying
    Via: SIP/2.0/WS 127.0.0.1:5166;branch=z9hG4bK214a3abf;rport
    To: <sip:q2uv843c@pi278aeds572.invalid;transport=ws>;tag=omq79k93vm
    From: "79999999999" <sip:101@127.0.0.1:5166>;tag=as5bdd7a38
    Call-ID: 45c330626f23227e638200ef0fefc8c6@127.0.0.1:5166
    CSeq: 103 INVITE
    Supported: outbound
    User-Agent: SIP.js/0.7.8
    Content-Length: 0
    
    
    
    2019/09/08 23:19:49.940433 127.0.0.1:8088 -> 127.0.0.1:59386
    ACK sip:q2uv843c@pi278aeds572.invalid;transport=ws SIP/2.0
    Via: SIP/2.0/WS 127.0.0.1:5166;branch=z9hG4bK67655c18;rport
    Max-Forwards: 70
    From: "79999999999" <sip:101@127.0.0.1:5166>;tag=as5bdd7a38
    To: <sip:q2uv843c@pi278aeds572.invalid;transport=ws>;tag=omq79k93vm
    Contact: <sip:101@127.0.0.1:5166;transport=ws>
    Call-ID: 45c330626f23227e638200ef0fefc8c6@127.0.0.1:5166
    CSeq: 103 ACK
    User-Agent: server
    Content-Length: 0
    
    
    
    2019/09/08 23:19:58.139024 127.0.0.1:8088 -> 127.0.0.1:59386
    BYE sip:q2uv843c@pi278aeds572.invalid;transport=ws SIP/2.0
    Via: SIP/2.0/WS 127.0.0.1:5166;branch=z9hG4bK37e054da;rport
    Max-Forwards: 70
    From: "79999999999" <sip:101@127.0.0.1:5166>;tag=as5bdd7a38
    To: <sip:q2uv843c@pi278aeds572.invalid;transport=ws>;tag=omq79k93vm
    Call-ID: 45c330626f23227e638200ef0fefc8c6@127.0.0.1:5166
    CSeq: 104 BYE
    User-Agent: server
    X-Asterisk-HangupCause: Normal Clearing
    X-Asterisk-HangupCauseCode: 16
    Content-Length: 0
    
    
    
    2019/09/08 23:19:58.203982 127.0.0.1:59386 -> 127.0.0.1:8088
    SIP/2.0 200 OK
    Via: SIP/2.0/WS 127.0.0.1:5166;branch=z9hG4bK37e054da;rport
    To: <sip:q2uv843c@pi278aeds572.invalid;transport=ws>;tag=omq79k93vm
    From: "79999999999" <sip:101@127.0.0.1:5166>;tag=as5bdd7a38
    Call-ID: 45c330626f23227e638200ef0fefc8c6@127.0.0.1:5166
    CSeq: 104 BYE
    Supported: outbound
    User-Agent: SIP.js/0.7.8
    Content-Length: 0