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Published 20 hours ago • verified publisher icon livekit

Closing room, BYE To Participant fails - 481 Call/Transaction Does Not Exist

Open ftsef opened this issue 8 months ago • 3 comments

Hello, I’ve discovered an issue that a BYE send from the LIVEKIT-SIP Server is not recognized by Asterisk. The SIP log showed a 481 Call/Transaction Does Not Exist.

It's reproducable on multiple Systems by on of the following steps:

  • removing a SIP-participant from a room
  • closing/deleteing a Room with a SIP-participant in it

The SIP participant will not be hung up and stays connected (to the void). I've used the latest docker image as well as build from master branch.

I'm going to check the RFC specifications but I thing it has something to do with the missing FROM tag in the BYE package.

Do you have a hint what it could be?

Thanks in advance.

<--- Received SIP request (923 bytes) from UDP:192.168.178.21:57881 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.178.21:57881;branch=z9hG4bK.u4EqNs7xF;rport
From: "1108" <sip:[email protected]>;tag=4c95ddtPY
To: sip:[email protected]
CSeq: 20 INVITE
Call-ID: sWT0ND51UG
Max-Forwards: 70
Supported: replaces, outbound, gruu, path, record-aware
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 300
Contact: <sip:[email protected]:57881;transport=udp>;expires=3599
User-Agent: Linphone-Desktop/5.2.4 (MBP.fritz.box) osx/14.5 Qt/5.15.2 LinphoneSDK/5.3.41

v=0
o=1108 3224 3649 IN IP4 192.168.178.21
s=Talk
c=IN IP4 192.168.178.21
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
a=record:off
m=audio 56905 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=rtcp:51058
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr

<--- Transmitting SIP response (482 bytes) to UDP:192.168.178.21:57881 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.178.21:57881;rport=57881;received=192.168.178.21;branch=z9hG4bK.u4EqNs7xF
Call-ID: sWT0ND51UG
From: "1108" <sip:[email protected]>;tag=4c95ddtPY
To: <sip:[email protected]>;tag=z9hG4bK.u4EqNs7xF
CSeq: 20 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1718736342/d004502f6d285bc24f0a1c981e36686f",opaque="7811bc271253cad3",algorithm=MD5,qop="auth"
Server: Asterisk PBX GIT-master-ea3b520bed
Content-Length:  0


<--- Received SIP request (336 bytes) from UDP:192.168.178.21:57881 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.178.21:57881;branch=z9hG4bK.u4EqNs7xF;rport
Call-ID: sWT0ND51UG
From: "1108" <sip:[email protected]>;tag=4c95ddtPY
To: <sip:[email protected]>;tag=z9hG4bK.u4EqNs7xF
Contact: <sip:[email protected]:57881;transport=udp>;expires=3599
Max-Forwards: 70
CSeq: 20 ACK


<--- Received SIP request (1202 bytes) from UDP:192.168.178.21:57881 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.178.21:57881;branch=z9hG4bK.Gg81mw2HT;rport
From: "1108" <sip:[email protected]>;tag=4c95ddtPY
To: sip:[email protected]
CSeq: 21 INVITE
Call-ID: sWT0ND51UG
Max-Forwards: 70
Supported: replaces, outbound, gruu, path, record-aware
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
Content-Type: application/sdp
Content-Length: 300
Contact: <sip:[email protected]:57881;transport=udp>;expires=3599
User-Agent: Linphone-Desktop/5.2.4 (MBP.fritz.box) osx/14.5 Qt/5.15.2 LinphoneSDK/5.3.41
Authorization:  Digest realm="asterisk", nonce="1718736342/d004502f6d285bc24f0a1c981e36686f", algorithm=MD5, opaque="7811bc271253cad3", username="1108",  uri="sip:[email protected]", response="59dbf5c01fdd269f4e55c4829eb55d83", cnonce="loefkneFJoVy9DSW", nc=00000001, qop=auth

v=0
o=1108 3224 3649 IN IP4 192.168.178.21
s=Talk
c=IN IP4 192.168.178.21
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
a=record:off
m=audio 56905 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=rtcp:51058
a=rtcp-fb:* trr-int 1000
a=rtcp-fb:* ccm tmmbr

<--- Transmitting SIP response (308 bytes) to UDP:192.168.178.21:57881 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.178.21:57881;rport=57881;received=192.168.178.21;branch=z9hG4bK.Gg81mw2HT
Call-ID: sWT0ND51UG
From: "1108" <sip:[email protected]>;tag=4c95ddtPY
To: <sip:[email protected]>
CSeq: 21 INVITE
Server: Asterisk PBX GIT-master-ea3b520bed
Content-Length:  0


    -- Executing [100@livekit:1] NoOp("PJSIP/1108-000004ce", "") in new stack
    -- Executing [100@livekit:2] Dial("PJSIP/1108-000004ce", "PJSIP/livekit") in new stack
    -- Called PJSIP/livekit
<--- Transmitting SIP request (951 bytes) to UDP:156.156.156.156:5060 --->
INVITE sip:+1234567890@testserver:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.42:5060;rport;branch=z9hG4bKPjbed0c2c0-e22d-44a2-91c0-2d7d4f54a284
From: "1108" <sip:[email protected]>;tag=d3cd97c4-5ffd-424d-a37c-e2a7597cd274
To: <sip:+1234567890@testserver>
Contact: <sip:[email protected]:5060>
Call-ID: 0f1f0682-90d5-4a62-b54e-e2a8cbfe0086
CSeq: 12643 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 GIT-master-ea3b520bed
Content-Type: application/sdp
Content-Length:   239

v=0
o=- 626213718 626213718 IN IP4 192.168.178.42
s=Asterisk
c=IN IP4 192.168.178.42
t=0 0
m=audio 22188 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Received SIP response (666 bytes) from UDP:156.156.156.156:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.178.42:5060;rport;branch=z9hG4bKPjbed0c2c0-e22d-44a2-91c0-2d7d4f54a284
From: "1108" <sip:[email protected]:5060>;tag=d3cd97c4-5ffd-424d-a37c-e2a7597cd274
To: <sip:+1234567890@testserver>;tag=ab420917-013e-4c75-9006-7c0b032d3b2e
Call-ID: 0f1f0682-90d5-4a62-b54e-e2a8cbfe0086
CSeq: 12643 INVITE
Content-Length:   232
Contact: <sip:156.156.156.156:5060>
Content-Type: application/sdp

v=0
o=- 626213718 626213720 IN IP4 156.156.156.156
s=LiveKit
c=IN IP4 156.156.156.156
t=0 0
m=audio 14510 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

       > 0x7fff0c050440 -- Strict RTP learning after remote address set to: 156.156.156.156:14510
<--- Transmitting SIP request (443 bytes) to UDP:156.156.156.156:5060 --->
ACK sip:156.156.156.156:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.42:5060;rport;branch=z9hG4bKPj46ddddc3-d282-4767-867c-87dc62ff8734
From: "1108" <sip:[email protected]>;tag=d3cd97c4-5ffd-424d-a37c-e2a7597cd274
To: <sip:+1234567890@testserver>;tag=ab420917-013e-4c75-9006-7c0b032d3b2e
Call-ID: 0f1f0682-90d5-4a62-b54e-e2a8cbfe0086
CSeq: 12643 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-master-ea3b520bed
Content-Length:  0


    -- PJSIP/livekit-000004cf answered PJSIP/1108-000004ce
       > 0x7fff0c0657d0 -- Strict RTP learning after remote address set to: 192.168.178.21:56905
<--- Transmitting SIP response (808 bytes) to UDP:192.168.178.21:57881 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.178.21:57881;rport=57881;received=192.168.178.21;branch=z9hG4bK.Gg81mw2HT
Call-ID: sWT0ND51UG
From: "1108" <sip:[email protected]>;tag=4c95ddtPY
To: <sip:[email protected]>;tag=13e525fa-cac5-480f-a338-2dd2de15664c
CSeq: 21 INVITE
Server: Asterisk PBX GIT-master-ea3b520bed
Contact: <sip:192.168.178.42:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   228

v=0
o=- 3224 3651 IN IP4 192.168.178.42
s=Asterisk
c=IN IP4 192.168.178.42
t=0 0
m=audio 5388 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv

    -- Channel PJSIP/livekit-000004cf joined 'simple_bridge' basic-bridge <3f4147bd-3ed0-472f-94a5-6d3d9a817817>
    -- Channel PJSIP/1108-000004ce joined 'simple_bridge' basic-bridge <3f4147bd-3ed0-472f-94a5-6d3d9a817817>
       > Bridge 3f4147bd-3ed0-472f-94a5-6d3d9a817817: switching from simple_bridge technology to native_rtp
       > Locally RTP bridged 'PJSIP/1108-000004ce' and 'PJSIP/livekit-000004cf' in stack
       > 0x7fff0c050440 -- Strict RTP switching to RTP target address 156.156.156.156:14510 as source
<--- Received SIP request (664 bytes) from UDP:192.168.178.21:57881 --->
ACK sip:192.168.178.42:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.21:57881;rport;branch=z9hG4bK.qDgHTsJk3
From: "1108" <sip:[email protected]>;tag=4c95ddtPY
To: <sip:[email protected]>;tag=13e525fa-cac5-480f-a338-2dd2de15664c
CSeq: 21 ACK
Call-ID: sWT0ND51UG
Max-Forwards: 70
Authorization:  Digest realm="asterisk", nonce="1718736342/d004502f6d285bc24f0a1c981e36686f", algorithm=MD5, opaque="7811bc271253cad3", username="1108",  uri="sip:[email protected]", response="59dbf5c01fdd269f4e55c4829eb55d83", cnonce="loefkneFJoVy9DSW", nc=00000001, qop=auth
User-Agent: Linphone-Desktop/5.2.4 (MBP.fritz.box) osx/14.5 Qt/5.15.2 LinphoneSDK/5.3.41


       > 0x7fff0c0657d0 -- Strict RTP switching to RTP target address 192.168.178.21:56905 as source
       > 0x7fff0c050440 -- Strict RTP learning complete - Locking on source address 156.156.156.156:14510
       > 0x7fff0c0657d0 -- Strict RTP learning complete - Locking on source address 192.168.178.21:56905
<--- Received SIP request (354 bytes) from UDP:156.156.156.156:5060 --->
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 123.123.123.123:5060;rport;branch=z9hG4bK.EycnBDy9dYjujMoL;alias
Max-Forwards: 70
Call-ID: 0f1f0682-90d5-4a62-b54e-e2a8cbfe0086
CSeq: 12644 BYE
Content-Length: 0
From: <sip:+1234567890@testserver>
To: "1108" <sip:[email protected]:5060>;tag=d3cd97c4-5ffd-424d-a37c-e2a7597cd274


<--- Transmitting SIP response (401 bytes) to UDP:156.156.156.156:5060 --->
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 123.123.123.123:5060;rport=5060;received=156.156.156.156;branch=z9hG4bK.EycnBDy9dYjujMoL;alias
Call-ID: 0f1f0682-90d5-4a62-b54e-e2a8cbfe0086
From: <sip:+1234567890@testserver>
To: "1108" <sip:[email protected]>;tag=d3cd97c4-5ffd-424d-a37c-e2a7597cd274
CSeq: 12644 BYE
Server: Asterisk PBX GIT-master-ea3b520bed
Content-Length:  0


<--- Transmitting SIP request (460 bytes) to UDP:192.168.178.21:57881 --->
OPTIONS sip:[email protected]:57881;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.178.42:5060;rport;branch=z9hG4bKPjaacda32c-e442-4391-a638-0e365bb80e33
From: <sip:[email protected]>;tag=ca840901-ac46-4bdb-908c-0098218166ef
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: a9352936-ad99-4080-8f73-17c43d9985f4
CSeq: 13343 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX GIT-master-ea3b520bed
Content-Length:  0


<--- Received SIP response (298 bytes) from UDP:192.168.178.21:57881 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.178.42:5060;rport;branch=z9hG4bKPjaacda32c-e442-4391-a638-0e365bb80e33
From: <sip:[email protected]>;tag=ca840901-ac46-4bdb-908c-0098218166ef
To: <sip:[email protected]>;tag=Y8rYb
Call-ID: a9352936-ad99-4080-8f73-17c43d9985f4
CSeq: 13343 OPTIONS

ftsef avatar Jun 19 '24 07:06 ftsef