[sr-dev] [kamailio/kamailio] Audio issue - STUN 401 Unauthorized with RTPEngine (#2567)
Slavik
notifications at github.com
Mon Nov 30 11:26:08 CET 2020
### Description
Hi,
I'm having an issue with outbound calls form my WebRTC client (SIP.JS via Web Socket). Results with no audio at all.
The issue is only happening for 1 call out of 3-5, and for some reason only on Google Chrome, on FireFox I am not able to reproduce it.
When the issue reproduced, what I'm seeing on Wireshark capture is that RTPEngine is sending me a STUN bind request, and I am sending back a response with error of 401 unauthorized (you can see it in the Wireshark capture).
After opening a similar issue on RTPEngine github:
https://github.com/sipwise/rtpengine/issues/1117
I understand that this may happen as a result of a wrong signaling path or something with the branches config - not really sure, as I don't yet fully understand how to work and troubleshoot those branches correctly.
**This is my deployment:**
- SIP.js 0.17.1 client.
- Kamailio 5.4.2
- RTPEgine 9.1.1.1 (installed on the Kamailio server itself)
- FreeSWITCH 1.10.5
- Google Chrome 87.0.4280.66
This is the flow:
WebRTC client ---(SIP via WSS)--> Kamailio (+RTPEngine) ---(SIP)--> FreeSWITCH ---(SIP)----> PSTN
### Troubleshooting
#### Reproduction
This is my whole config file:
https://gist.github.com/slavikbialik/f8b360184be31dbc55a3eb604ac6cb4e
Then, with my WebRTC client (SIP.js client) I'm making an outbound call towards PSTN.
#### Log Messages
The logs below are from both components Kamailio + RTPEngine.
```
13(663) INFO: <script>: START: INVITE from sip:1000 at userdomain.com (IP:192.168.2.183:55692)
13(663) INFO: <script>: MANAGE_BRANCH: New branch [0] to sip:+12126131234 at userdomain.com
13(663) INFO: <script>: NATMANAGE branch_id:0 ruri: sip:+12126131234 at userdomain.com, method:INVITE, status:<null>, extra_id: z9hG4bK1989630, rtpengine_manage: replace-origin replace-session-connection trust-address via-branch=extra rtcp-mux-demux generate-mid DTLS=off ICE=remove RTP/AVP
[1606730670.515178] INFO: [ji6h1tveo2ota7klijqn]: Received command 'offer' from 172.29.0.2:41365
[1606730670.515319] NOTICE: [ji6h1tveo2ota7klijqn]: Creating new call
[1606730670.516489] INFO: [ji6h1tveo2ota7klijqn]: Replying to 'offer' from 172.29.0.2:41365 (elapsed time 0.001287 sec)
[1606730670.518452] INFO: [ji6h1tveo2ota7klijqn port 23020]: ICE negotiated: peer for component 1 is 192.168.2.183:59040
[1606730670.518472] INFO: [ji6h1tveo2ota7klijqn port 23020]: ICE negotiated: local interface 172.29.0.2
14(664) INFO: <script>: BRANCH FAILED: z9hG4bK198963 + 0[1606730670.584982] INFO: [ji6h1tveo2ota7klijqn]: Received command 'delete' from 172.29.0.2:43389
[1606730670.585179] INFO: [ji6h1tveo2ota7klijqn]: Deleting call branch '' (via-branch 'z9hG4bK1989630')
[1606730670.585285] INFO: [ji6h1tveo2ota7klijqn]: Call branch '' (via-branch 'z9hG4bK1989630') deleted, no more branches remaining
[1606730670.585295] INFO: [ji6h1tveo2ota7klijqn]: Deleting entire call
[1606730670.585471] INFO: [ji6h1tveo2ota7klijqn]: Final packet stats:
[1606730670.585483] INFO: [ji6h1tveo2ota7klijqn]: --- Tag 'ij9um6cg76', created 0:00 ago for branch '', in dialogue with ''
[1606730670.585488] INFO: [ji6h1tveo2ota7klijqn]: ------ Media #1 (audio over UDP/TLS/RTP/SAVPF) using unknown codec
[1606730670.585493] INFO: [ji6h1tveo2ota7klijqn]: --------- Port 172.29.0.2:23020 <> 192.168.2.183:59040, SSRC 0, 1 p, 64 b, 0 e, 0 ts
[1606730670.585497] INFO: [ji6h1tveo2ota7klijqn]: --- Tag '', created 0:00 ago for branch 'z9hG4bK1989630', in dialogue with 'ij9um6cg76'
[1606730670.585504] INFO: [ji6h1tveo2ota7klijqn]: ------ Media #1 (audio over RTP/AVP) using unknown codec
[1606730670.585508] INFO: [ji6h1tveo2ota7klijqn]: --------- Port 172.29.0.2:23000 <> :0 , SSRC 0, 0 p, 0 b, 0 e, 0 ts
[1606730670.585613] INFO: [ji6h1tveo2ota7klijqn]: --------- Port 172.29.0.2:23001 <> :0 (RTCP), SSRC 0, 0 p, 0 b, 0 e, 0 ts
[1606730670.585812] INFO: [ji6h1tveo2ota7klijqn]: Replying to 'delete' from 172.29.0.2:43389 (elapsed time 0.000800 sec)
14(664) INFO: <script>: Failure: <null>13(663) INFO: <script>: START: ACK from sip:1000 at userdomain.com (IP:192.168.2.183:55692)
13(663) INFO: <script>: START: INVITE from sip:1000 at userdomain.com (IP:192.168.2.183:55692)
13(663) INFO: <script>: MANAGE_BRANCH: New branch [0] to sip:+12126131234 at userdomain.com
13(663) INFO: <script>: NATMANAGE branch_id:0 ruri: sip:+12126131234 at userdomain.com, method:INVITE, status:<null>, extra_id: z9hG4bK9886370, rtpengine_manage: replace-origin replace-session-connection trust-address via-branch=extra rtcp-mux-demux generate-mid DTLS=off ICE=remove RTP/AVP
[1606730670.593366] INFO: [ji6h1tveo2ota7klijqn]: Received command 'offer' from 172.29.0.2:41365
[1606730670.593703] NOTICE: [ji6h1tveo2ota7klijqn]: Creating new call
[1606730670.594492] INFO: [ji6h1tveo2ota7klijqn]: Replying to 'offer' from 172.29.0.2:41365 (elapsed time 0.000970 sec)
[1606730670.596288] NOTICE: [ji6h1tveo2ota7klijqn port 23020]: Received invalid STUN packet from 192.168.2.183:59040: MESSAGE_INTEGRITY attribute missing
14(664) INFO: <script>: NATMANAGE branch_id:0 ruri: <null>, method:INVITE, status:100, extra_id: 0, rtpengine_manage: replace-origin replace-session-connection trust-address via-branch=extra rtcp-mux-offer generate-mid DTLS=active SDES-off ICE=force RTP/SAVPF
15(665) INFO: <script>: START: NOTIFY from sip:1000 at userdomain.com (IP:172.29.0.1:45402)
15(665) INFO: <script>: START: NOTIFY from sip:1000 at userdomain.com (IP:172.29.0.1:45402)
14(664) INFO: <script>: NATMANAGE branch_id:0 ruri: <null>, method:INVITE, status:183, extra_id: z9hG4bK9886370, rtpengine_manage: replace-origin replace-session-connection via-branch=extra rtcp-mux-offer generate-mid DTLS=active SDES-off ICE=force RTP/SAVPF
[1606730671.014062] INFO: [ji6h1tveo2ota7klijqn]: Received command 'answer' from 172.29.0.2:43389
[1606730671.014417] INFO: [ji6h1tveo2ota7klijqn]: Replying to 'answer' from 172.29.0.2:43389 (elapsed time 0.000337 sec)
[1606730671.065289] ERR: [ji6h1tveo2ota7klijqn port 23000]: SRTP output wanted, but no crypto suite was negotiated
[1606730675.002358] INFO: [ji6h1tveo2ota7klijqn port 23000]: Confirmed peer address as 172.29.0.1:34505
14(664) INFO: <script>: NATMANAGE branch_id:0 ruri: <null>, method:INVITE, status:200, extra_id: z9hG4bK9886370, rtpengine_manage: replace-origin replace-session-connection via-branch=extra rtcp-mux-offer generate-mid DTLS=active SDES-off ICE=force RTP/SAVPF
[1606730675.504902] INFO: [ji6h1tveo2ota7klijqn]: Received command 'answer' from 172.29.0.2:43389
[1606730675.505259] INFO: [ji6h1tveo2ota7klijqn]: Replying to 'answer' from 172.29.0.2:43389 (elapsed time 0.000342 sec)
15(665) INFO: <script>: START: NOTIFY from sip:1000 at userdomain.com (IP:172.29.0.1:45402)
15(665) INFO: <script>: START: NOTIFY from sip:1000 at userdomain.com (IP:172.29.0.1:45402)
13(663) INFO: <script>: START: ACK from sip:1000 at userdomain.com (IP:192.168.2.183:55692)
13(663) INFO: <script>: NATMANAGE branch_id:0 ruri: sip:+12126131234 at 192.168.40.60:5081;transport=tcp, method:ACK, status:<null>, extra_id: <null>, rtpengine_manage: replace-origin replace-session-connection trust-address rtcp-mux-demux generate-mid DTLS=off ICE=remove RTP/AVP
13(663) INFO: <script>: START: SUBSCRIBE from sip:1000 at userdomain.com (IP:192.168.2.183:55692)
13(663) INFO: <script>: MANAGE_BRANCH: New branch [0] to sip:ji6h1tveo2ota7klijqn at userdomain.com
13(663) INFO: <script>: NATMANAGE branch_id:0 ruri: sip:ji6h1tveo2ota7klijqn at userdomain.com, method:SUBSCRIBE, status:<null>, extra_id: z9hG4bK53253160, rtpengine_manage: replace-origin replace-session-connection trust-address via-branch=extra rtcp-mux-demux generate-mid DTLS=off ICE=remove RTP/AVP
12(662) INFO: <script>: NATMANAGE branch_id:0 ruri: <null>, method:SUBSCRIBE, status:202, extra_id: z9hG4bK53253160, rtpengine_manage: replace-origin replace-session-connection trust-address via-branch=extra rtcp-mux-offer generate-mid DTLS=active SDES-off ICE=force RTP/SAVPF
12(662) INFO: <script>: START: NOTIFY from sip:ji6h1tveo2ota7klijqn at userdomain.com (IP:172.29.0.1:45434)
[1606730679.005129] INFO: [ji6h1tveo2ota7klijqn port 23000]: Confirmed peer address as 172.29.0.1:34505
14(664) INFO: <script>: START: NOTIFY from sip:ji6h1tveo2ota7klijqn at userdomain.com (IP:172.29.0.1:45434)
14(664) INFO: <script>: START: NOTIFY from sip:ji6h1tveo2ota7klijqn at userdomain.com (IP:172.29.0.1:45434)
15(665) INFO: <script>: START: BYE from sip:+12126131234 at userdomain.com (IP:172.29.0.1:45402)
15(665) INFO: <script>: MANAGE_BRANCH: New branch [0] to sip:gco78n18 at userdomain.com;transport=ws;ob;alias=192.168.2.183~55692~6
15(665) INFO: <script>: NATMANAGE branch_id:0 ruri: sip:gco78n18 at userdomain.com;transport=ws;ob;alias=192.168.2.183~55692~6, method:BYE, status:<null>, extra_id: <null>, rtpengine_manage: replace-origin replace-session-connection trust-address rtcp-mux-offer generate-mid DTLS=active SDES-off ICE=force RTP/SAVPF
[1606730685.093720] INFO: [ji6h1tveo2ota7klijqn]: Received command 'delete' from 172.29.0.2:58025
[1606730685.093800] INFO: [ji6h1tveo2ota7klijqn]: Deleting call branch '9ZpZKU852jUrg' (via-branch '')
[1606730685.093811] INFO: [ji6h1tveo2ota7klijqn]: Call branch '9ZpZKU852jUrg' (via-branch 'z9hG4bK9886370') deleted, no more branches remaining
[1606730685.093816] INFO: [ji6h1tveo2ota7klijqn]: Deleting entire call
[1606730685.093820] INFO: [ji6h1tveo2ota7klijqn]: Final packet stats:
[1606730685.093824] INFO: [ji6h1tveo2ota7klijqn]: --- Tag 'ij9um6cg76', created 0:15 ago for branch '', in dialogue with '9ZpZKU852jUrg'
[1606730685.093830] INFO: [ji6h1tveo2ota7klijqn]: ------ Media #1 (audio over RTP/SAVPF) using unknown codec
[1606730685.093838] INFO: [ji6h1tveo2ota7klijqn]: --------- Port 172.29.0.2:23020 <> 82.166.70.46:59040, SSRC 0, 0 p, 0 b, 0 e, 15 ts
[1606730685.093843] INFO: [ji6h1tveo2ota7klijqn]: --- Tag '9ZpZKU852jUrg', created 0:15 ago for branch 'z9hG4bK9886370', in dialogue with 'ij9um6cg76'
[1606730685.093852] INFO: [ji6h1tveo2ota7klijqn]: ------ Media #1 (audio over RTP/AVP) using PCMU/8000
[1606730685.093858] INFO: [ji6h1tveo2ota7klijqn]: --------- Port 172.29.0.2:23000 <> 172.29.0.1:34505, SSRC bbc97896, 692 p, 119024 b, 0 e, 0 ts
[1606730685.093865] INFO: [ji6h1tveo2ota7klijqn]: --------- Port 172.29.0.2:23001 <> 192.168.40.60:20045 (RTCP), SSRC 0, 0 p, 0 b, 0 e, 15 ts
```
As you can see in the logs, for every call I'm reaching this error (although not every call has the issue):
`14(664) INFO: <script>: BRANCH FAILED: z9hG4bK198963 + 0[1606730670.584982] INFO: [ji6h1tveo2ota7klijqn]: Received command 'delete' from 172.29.0.2:43389`
And ONLY when issue occurs we can see this log from RTPEngine:
`[1606730670.596288] NOTICE: [ji6h1tveo2ota7klijqn port 23020]: Received invalid STUN packet from 192.168.2.183:59040: MESSAGE_INTEGRITY attribute missing`
What I was able to understand (thanks to @rfuchs from RTPEngine from the issue I opened over there) is that is because of `rtpengine_delete` command invoked, and the only place it is invoked is right in this code block:
```
event_route[tm:branch-failure:rtpengine] {
xlog("L_INFO", "BRANCH FAILED: $sel(via[1].branch) + $T_branch_idx");
#!ifdef WITH_BRIDGE_ON_FAIL
if (!isbflagset(FLB_BRIDGE) && t_check_status("415|488")) {
t_reuse_branch();
setbflag(FLB_BRIDGE);
xlog("L_INFO", "event_route[branch-failure:rtpengine]: trying again\n");
route(RELAY);
} else {
$avp(extra_id) = @via[1].branch + $T_branch_idx;
rtpengine_delete("via-branch=extra");
xlog("L_INFO", "event_route[branch-failure:rtpengine]: failed\n");
}
#!else
$avp(extra_id) = @via[1].branch + $T_branch_idx;
rtpengine_delete("via-branch=extra");
#!endif
}
```
Unfortunately, I don't really understand when this event route is being invoked and why it is invoked. Because if so, maybe I can solve the root issue so there won't be a branch failure that will lead to this `rtpengine_delete` command to be invoked.
And tell you the truth, I'm not really sure what this code block does at all. 🤔 But I do understand that probably it should be there. All calls are getting to this code block, because all calls are printing the error in this code block.
#### SIP Traffic
I have 2 captures included in the ZIP file.
1. My PC that is running the WebRTC client (filtered with stun and dtls)
2. Kamailio (which also has the RTPEngine on it)
[stun-issue-no-audio.zip](https://github.com/kamailio/kamailio/files/5615289/stun-issue-no-audio.zip)
Also here's an output of the SIP (INVITE + 183 + 200) from my WebRTC client:
```
INVITE sip:+12126131234 at userdomain.com SIP/2.0
Via: SIP/2.0/WSS userdomain.com;branch=z9hG4bK988637
To: <sip:+12126131234 at userdomain.com>
From: <sip:1000 at userdomain.com>;tag=ij9um6cg76
CSeq: 2 INVITE
Call-ID: ji6h1tveo2ota7klijqn
Max-Forwards: 70
Proxy-Authorization: Digest algorithm=MD5, username="1000", realm="userdomain.com", nonce="2588da22-d5a2-424c-9086-495e5a21e98d", uri="sip:+12126131234 at userdomain.com", response="16b532e9d6d111316ee3ab46555398f6", qop=auth, cnonce="acfjkcjsgeis", nc=00000001
Contact: <sip:gco78n18 at userdomain.com;transport=ws;ob>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: SIP.js/0.17.1
Content-Type: application/sdp
Content-Length: 1981
v=0
o=- 8504452169629147650 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=msid-semantic: WMS ad2b7e59-f25c-492e-9aa9-e03fe86d09f9
m=audio 59040 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 82.166.70.46
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:4020639172 1 udp 2122260223 172.18.48.1 59039 typ host generation 0 network-id 1
a=candidate:535800724 1 udp 2122194687 192.168.2.183 59040 typ host generation 0 network-id 2
a=candidate:2661788960 1 udp 1685987071 82.166.70.46 59040 typ srflx raddr 192.168.2.183 rport 59040 generation 0 network-id 2
a=ice-ufrag:jevk
a=ice-pwd:6bWdl0Gtcgen9eumdaiajcFJ
a=ice-options:trickle
a=fingerprint:sha-256 73:E3:2C:B6:5D:BC:E9:1C:40:AB:FB:34:F2:A4:9D:65:94:B9:DA:07:1C:16:BB:67:29:16:8C:0D:3A:84:6B:35
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:ad2b7e59-f25c-492e-9aa9-e03fe86d09f9 f33e36f5-a06a-4379-8e03-017811d23781
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:2583426934 cname:fzLi/Lg2BSxZuSFA
a=ssrc:2583426934 msid:ad2b7e59-f25c-492e-9aa9-e03fe86d09f9 f33e36f5-a06a-4379-8e03-017811d23781
a=ssrc:2583426934 mslabel:ad2b7e59-f25c-492e-9aa9-e03fe86d09f9
a=ssrc:2583426934 label:f33e36f5-a06a-4379-8e03-017811d23781
SIP/2.0 183 Session Progress
Via: SIP/2.0/WSS userdomain.com;rport=55692;received=192.168.2.183;branch=z9hG4bK988637
Record-Route: <sip:192.168.40.60:5060;transport=tcp;r2=on;lr=on;ftag=ij9um6cg76;rtp=bridge;rtp=ws>
Record-Route: <sip:0.0.0.0:8443;transport=ws;r2=on;lr=on;ftag=ij9um6cg76;rtp=bridge;rtp=ws>
From: <sip:1000 at userdomain.com>;tag=ij9um6cg76
To: <sip:+12126131234 at userdomain.com>;tag=9ZpZKU852jUrg
Call-ID: ji6h1tveo2ota7klijqn
CSeq: 2 INVITE
Contact: <sip:+12126131234 at 192.168.40.60:5081;transport=tcp>
User-Agent: FreeSWITCH-mod_sofia/1.10.5-release+git~20200818T185121Z~25569c1631~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 586
Remote-Party-ID: "Outbound Call" <sip:+12126131234 at userdomain.com>;party=calling;privacy=off;screen=no
v=0
o=FreeSWITCH 1606710627 1606710628 IN IP4 192.168.40.60
s=FreeSWITCH
c=IN IP4 192.168.40.60
t=0 0
m=audio 23020 RTP/SAVPF 0 126
a=mid:0
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=sendrecv
a=rtcp:23020
a=rtcp-mux
a=setup:active
a=fingerprint:sha-256 DD:04:45:36:F9:BD:66:7A:4F:7A:5A:E3:1A:2A:C1:E2:16:F7:AA:F5:89:B3:74:73:72:D8:1D:C4:97:63:6C:79
a=ptime:20
a=ice-ufrag:OANWcJ8l
a=ice-pwd:QLO3QVUf7YqfR5PLmCA2zqKUK0
a=ice-options:trickle
a=candidate:8DY2z151lUdw24NX 1 UDP 2130706431 192.168.40.60 23020 typ host
a=end-of-candidates
SIP/2.0 200 OK
Via: SIP/2.0/WSS userdomain.com;rport=55692;received=192.168.2.183;branch=z9hG4bK988637
Record-Route: <sip:192.168.40.60:5060;transport=tcp;r2=on;lr=on;ftag=ij9um6cg76;rtp=bridge;rtp=ws>
Record-Route: <sip:0.0.0.0:8443;transport=ws;r2=on;lr=on;ftag=ij9um6cg76;rtp=bridge;rtp=ws>
From: <sip:1000 at userdomain.com>;tag=ij9um6cg76
To: <sip:+12126131234 at userdomain.com>;tag=9ZpZKU852jUrg
Call-ID: ji6h1tveo2ota7klijqn
CSeq: 2 INVITE
Contact: <sip:+12126131234 at 192.168.40.60:5081;transport=tcp>
User-Agent: FreeSWITCH-mod_sofia/1.10.5-release+git~20200818T185121Z~25569c1631~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 586
X-Conference-Name: ji6h1tveo2ota7klijqn at 192.168.40.60:5071
Remote-Party-ID: "Outbound Call" <sip:+12126131234 at userdomain.com>;party=calling;privacy=off;screen=no
v=0
o=FreeSWITCH 1606710627 1606710628 IN IP4 192.168.40.60
s=FreeSWITCH
c=IN IP4 192.168.40.60
t=0 0
m=audio 23020 RTP/SAVPF 0 126
a=mid:0
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=sendrecv
a=rtcp:23020
a=rtcp-mux
a=setup:active
a=fingerprint:sha-256 DD:04:45:36:F9:BD:66:7A:4F:7A:5A:E3:1A:2A:C1:E2:16:F7:AA:F5:89:B3:74:73:72:D8:1D:C4:97:63:6C:79
a=ptime:20
a=ice-ufrag:OANWcJ8l
a=ice-pwd:QLO3QVUf7YqfR5PLmCA2zqKUK0
a=ice-options:trickle
a=candidate:8DY2z151lUdw24NX 1 UDP 2130706431 192.168.40.60 23020 typ host
a=end-of-candidates
```
### Possible Solutions
The only workaround I found is to delete the `rtpengine_delete` command invoked in the event route. But then it is not solving the root issue of my problem, and then I also experiencing a delay of 1-2 seconds until audio starts.
### Additional Information
* **Kamailio Version** - output of `kamailio -v`
```
version: kamailio 5.4.2 (x86_64/linux) c3b91f
flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: c3b91f
compiled on 16:08:56 Nov 25 2020 with gcc 8.3.0
```
* **Operating System**:
```
Debian 10.6
Linux kamailio 4.19.0-8-amd64 #1 SMP Debian 4.19.98-1 (2020-01-26) x86_64 GNU/Linux
```
Please let me know if you need me to add any other logs or information.
Thank you!
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https://github.com/kamailio/kamailio/issues/2567
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