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Revision 2
grahamb's avatar
23.8k
grahamb
updated 2021-08-17 13:28:57 +0000

Cannot see SIP call flow

Hi, I am using WebRTC to make a call between a SipPhone and a Browser.

  • When the phone is ringing, there is no INVITE in WireShark
  • While talking, this repeats: protocol: UDP | length: 214 | Info: 31410 -> 9014 Len:172
  • When I end the call: protocol: SIP | length: 509 | Info: Request: BYE sip:[email protected]:5060 |

I really appreciate your help. Thank you

Revision 1
Nanando's avatar
1
Nanando
asked 2021-08-17 13:26:04 +0000

Cannot see SIP call flow

Hi, I am using WebRTC to make a call between a SipPhone and a Browser.

  • Browser. - When the phone is ringing, there is no INVITE in WireShark
  • WireShark - While talking, this repeats: protocol: UDP | length: 214 | Info: 31410 -> 9014 Len:172
  • Len:172 - When I end the call: protocol: SIP | length: 509 | Info: Request: BYE sip:[email protected]:5060 |

I really appreciate your help. Thank you