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Beginner
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SIP issue

Hi,

I encounter some problems if I create a room with SIP connectivity and in the session there is also a Mac client connected. Some users in the room do not receive any packets on the SIP stream even if they have successfully subscribed it.

I can not find any significant errors or warnings in the intelcs logs.

If in the session are present only windows or linux clients, the problem does not occur and everything works correctly.

Do you have any idea of ​​what the problem might be?

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Beginner
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Hi,

Hi,

I integrate my previous post. I have identified this error in the logs:

 

2018-01-08 15:59:04.974  - DEBUG: AmqpClient - New message received { method: 'createInternalConnection',
 args: [ '5fe592f9e05f', 'in', { protocol: 'tcp' } ],
 corrID: 622,
 replyTo: 'amq.gen-NLUbKnl_G470LPwX5NhW5w' }
2018-01-08 15:59:04.978  - DEBUG: AmqpClient - New message received { method: 'linkup',
 args: [ 'Ias3I4kjRINjUu3KAATY-sub-5fe592f9e05f', '5fe592f9e05f', null ],
 corrID: 624,
 replyTo: 'amq.gen-NLUbKnl_G470LPwX5NhW5w' }
2018-01-08 15:59:04.978  - DEBUG: Connections - linkup, connectionId: Ias3I4kjRINjUu3KAATY-sub-5fe592f9e05f , audioFrom: 5fe592f9e05f , videoFrom: null
2018-01-08 15:59:04.978  - ERROR: Connections - Audio stream does not exist:5fe592f9e05f
2018-01-08 15:59:04.989  - DEBUG: AmqpClient - New message received { method: 'publish',
 args:
  [ '5fe592f9e05f',
    'internal',
    { controller: 'session.dcfc4a54-012d-a61d-39ab-8853310949c2.323',
      owner: 'sip:
xxxxxxxxxxxx@xxx.xxx.xxx.xxx',
      audio: [Object],
      video: false,
      ip: 'xxx.xxx.xxx.xxx',
      port: 0 } ],
 corrID: 625,
 replyTo: 'amq.gen-NLUbKnl_G470LPwX5NhW5w' }
2018-01-08 15:59:04.990  - DEBUG: WebrtcNode - publish, connectionId: 5fe592f9e05f connectionType: internal options: { controller: 'session.dcfc4a54-012d-a61d-39ab-8853310949c2.323',
 owner: 'sip:
xxxxxxxxxxxx@xxx.xxx.xxx.xxx',
 audio: { codec: 'opus_48000_2' },
 video: false,
 ip: '
xxx.xxx.xxx.xxx',
 port: 0 }

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Beginner
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Hi, Any updates?

Hi,

Any updates?

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Beginner
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Hi team,

Hi team,

if I delay the subscription of the sip stream with a setTimeout, it seems that the problem does not occur.

It seems that when I receive the "stream-added" event, the sip audio stream is not yet ready (see above webrtc-agent log file).

This only happens if I subscribe to a sip stream. 

 

 

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