sip icon indicating copy to clipboard operation
sip copied to clipboard

Wrong connection IP address in 200 SDP

Open EdiMel opened this issue 1 year ago • 1 comments

Hi.. I'm calling from an asterisk's sip trunk but have no voice in one side. Here is the invite that asterisk send to live kit:

2024/08/15 14:09:01.576435 172.31.32.218:5060 -> 129.213.105.135:5060
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;rport;branch=z9hG4bKPj95ed0026-fbda-49c4-bb43-b4311ac142e7
From: <sip:[email protected]>;tag=147ad562-9247-4679-a30b-bfbcbb1628a4
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: 7837c9d0-cd7c-4899-906d-4420919e5913
CSeq: 28856 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.9.2
Content-Type: application/sdp
Content-Length:   353

v=0
o=- 1145153493 1145153493 IN IP4 xxx.xxx.xxx.xxx
s=Asterisk
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 14680 RTP/AVP 107 0 101 102
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/48000
a=fmtp:101 0-16
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

This is the 180:

2024/08/15 14:09:01.593919 129.213.105.135:5060 -> 172.31.32.218:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;rport;branch=z9hG4bKPj95ed0026-fbda-49c4-bb43-b4311ac142e7
From: <sip:[email protected]>;tag=147ad562-9247-4679-a30b-bfbcbb1628a4
To: <sip:[email protected]>;tag=118ea5a8-354b-4762-97ea-f302c5b728af
Call-ID: 7837c9d0-cd7c-4899-906d-4420919e5913
CSeq: 28856 INVITE
Content-Length: 0

And this is the 200 (from a different IP than the 100 and 180):

2024/08/15 14:09:01.608672 150.136.76.146:9000 -> 172.31.32.218:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;rport;branch=z9hG4bKPj95ed0026-fbda-49c4-bb43-b4311ac142e7
From: <sip:[email protected]>;tag=147ad562-9247-4679-a30b-bfbcbb1628a4
To: <sip:[email protected]>;tag=9259a4fd-4a12-4f04-b17a-8cdc8b44fbd4
Call-ID: 7837c9d0-cd7c-4899-906d-4420919e5913
CSeq: 28856 INVITE
Content-Length: 218
Contact: <sip:10.0.134.47:9000>
Content-Type: application/sdp

v=0
o=- 1145153493 1145153495 IN IP4 10.0.134.47
s=LiveKit
c=IN IP4 10.0.134.47
t=0 0
m=audio 54152 RTP/AVP 0 102
a=rtpmap:0 PCMU/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=sendrecv

As you can see, its using the internal IP from contact and SDP connection info.

EdiMel avatar Aug 15 '24 14:08 EdiMel

Additional info for debugging: callID=SCL_TscmohvBXLWu.

dennwc avatar Aug 15 '24 17:08 dennwc

Should be fixed now.

dennwc avatar Aug 22 '24 15:08 dennwc