Cannot see SIP call flow

Nanando
1 ●1 ●1

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

1 Answer

Sort by » oldest newest most voted

BigFatCat
31 ●3 ●90 ●5

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.

Please start posting anonymously - your entry will be published after you log in or create a new account.

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

Seen: 551 times

Last updated: Aug 18 '21