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Cannot see SIP call flow

asked 2021-08-17 13:26:04 +0000

updated 2021-08-17 13:28:57 +0000

grahamb gravatar image

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

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answered 2021-08-18 08:36:24 +0000

BigFatCat gravatar image
  • When the phone is ringing, there is no INVITE in WireShark. The INVITE is not for ringing. It is typically 180 or 183.
  • UDP | length: 214 | Info: 31410 -> 9014 Len:172 is most likely RTP, G.711 packets. Need to verify if Wireshark UDP ports 31410 and 9014 is decode as "RTP"
  • When I end the call: protocol: SIP | length: 509 | Info: Request: BYE sip:[email protected]:5060 I am not sure what the 1000 means. The 192.168.111.143:5060 is the port (5060) for SIP. Verify the information in the SIP header if it was your test call then filter by Call-ID.
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Asked: 2021-08-17 13:26:04 +0000

Seen: 71 times

Last updated: Aug 18