Response msg 404/INVITE error when calling a doorbell

sesardelaisla · November 20, 2024, 2:01pm

Hello Everyone,

I am getting a 404 error when calling to a HikVision doorbell which is correctly registered in Asterisk. Actually, I have two doorbells: they are the same model but different hardware and firmware version (one is older than the other one). The one I am experiencing issues is with the older one. The newer one works fine. I have done the tests with the same extension number (not connected at the same time, of course) to be sure that the Asterisk installation or extension config are not the problem.

I guess there is some sort of protocol implementation difference in the older doorbell, but I want to be sure that I am not missing any thing, or to check whether there is any workaround for both doorbells to work on some sort of “legacy” workflow. Pasting below the logs for the working doorbell (which issues a 200 status prior to making the call; call is established and everything works fine) and the one which issues the 404 error and a busy/congestion message:

[Using pastebin due to forum character count limit]

200

404

I think the first related error is in the following line:

[Nov 20 13:45:13] DEBUG[722]: res_pjsip/pjsip_distributor.c:499 distributor: Searching for serializer associated with dialog dlg0x7f55b8010a68 for Response msg 404/INVITE/cseq=20502 (rdata0x7f55d0002b58)

I don’t find another relevant difference in the 200 response given by the “good” doorbell, so I guess that’s when the problem starts to be displayed in the log.

Any advice, please? Happy to share further information you guys might need. Thanks.

Sésar

jcolp · November 20, 2024, 2:16pm

That’s not an error. That’s a debug message. If you turn on debug and don’t understand things, you’ll think lots of stuff are errors when they are perfectly normal and expected.

You haven’t included the actual SIP traffic, which is needed.

sesardelaisla · November 20, 2024, 2:27pm

I will figure out how to get SIP traffic. I think I did it in the past but I don’t remember how. My Asterisk instance is an addon installed in the Home Assistant automation system, so it is not a standard setup. Will give further feedback asap. Thanks.

sesardelaisla · November 21, 2024, 11:29am

Is this what you were expecting me to share?

GOOD DOORBELL (Registration and call to doorbell - 111)

No.   Timestamp  (Dir) Address                  SIP Message                        
===== ========== ============================== ===================================
00000 1732188242 * ==> 10.31.255.134:5070       OPTIONS sip:949xxxxxx@telefonica.net SIP/2.0
00001 1732188242 * <== 10.31.255.134:5070       OPTIONS sip:949xxxxxx@192.168.0.100:5060 SIP/2.0
00002 1732188242 * ==> 10.31.255.134:5070       SIP/2.0 404 Not Found
00003 1732188242 * <== 10.31.255.134:5070       SIP/2.0 200 OK
00004 1732188247 * <== 192.168.0.72:5060        REGISTER sip:192.168.0.100:5060 SIP/2.0
00005 1732188247 * ==> 192.168.0.72:5060        SIP/2.0 401 Unauthorized
00006 1732188247 * <== 192.168.0.72:5060        REGISTER sip:192.168.0.100:5060 SIP/2.0
00007 1732188247 * ==> 192.168.0.72:5060        SIP/2.0 200 OK
00008 1732188250 * ==> 10.31.255.134:5070       REGISTER sip:10.31.255.134:5070 SIP/2.0
00009 1732188250 * <== 10.31.255.134:5070       SIP/2.0 200 OK
00010 1732188258 * <== 192.168.0.1:58892        INVITE sip:111@ha.MY_DOMAIN.com SIP/2.0
00011 1732188258 * ==> 192.168.0.1:58892        SIP/2.0 401 Unauthorized
00012 1732188258 * <== 192.168.0.1:58892        ACK sip:111@ha.MY_DOMAIN.com SIP/2.0
00013 1732188258 * <== 192.168.0.1:58892        INVITE sip:111@ha.MY_DOMAIN.com SIP/2.0
00014 1732188258 * ==> 192.168.0.1:58892        SIP/2.0 100 Trying
00015 1732188258 * ==> 192.168.0.72:5060        INVITE sip:111@192.168.0.72:5060 SIP/2.0
00016 1732188258 * <== 192.168.0.72:5060        SIP/2.0 100 Trying
00017 1732188258 * <== 192.168.0.72:5060        SIP/2.0 200 OK
00018 1732188258 * ==> 192.168.0.72:5060        ACK sip:111@192.168.0.72:5060 SIP/2.0
00019 1732188258 * ==> 192.168.0.1:58892        SIP/2.0 200 OK
00020 1732188258 * <== 192.168.0.1:58892        ACK sip:192.168.0.100:18089;transport=ws SIP/2.0
00021 1732188268 * <== 192.168.0.1:58892        BYE sip:192.168.0.100:18089;transport=ws SIP/2.0
00022 1732188268 * ==> 192.168.0.1:58892        SIP/2.0 200 OK
00023 1732188268 * ==> 192.168.0.72:5060        BYE sip:111@192.168.0.72:5060 SIP/2.0
00024 1732188268 * <== 192.168.0.72:5060        SIP/2.0 100 Trying
00025 1732188268 * <== 192.168.0.72:5060        SIP/2.0 200 OK

BAD DOORBELL (Registration and call to doorbell - 111)

No.   Timestamp  (Dir) Address                  SIP Message                        
===== ========== ============================== ===================================
00000 1732188171 * <== 192.168.0.222:5060       REGISTER sip:192.168.0.100:5060 SIP/2.0
00001 1732188171 * ==> 192.168.0.222:5060       SIP/2.0 401 Unauthorized
00002 1732188171 * <== 192.168.0.222:5060       REGISTER sip:192.168.0.100:5060 SIP/2.0
00003 1732188171 * ==> 192.168.0.222:5060       SIP/2.0 200 OK
00004 1732188182 * ==> 10.31.255.134:5070       OPTIONS sip:949xxxxxx@telefonica.net SIP/2.0
00005 1732188182 * <== 10.31.255.134:5070       OPTIONS sip:949xxxxxx@192.168.0.100:5060 SIP/2.0
00006 1732188182 * ==> 10.31.255.134:5070       SIP/2.0 404 Not Found
00007 1732188182 * <== 10.31.255.134:5070       SIP/2.0 200 OK
00008 1732188190 * ==> 10.31.255.134:5070       REGISTER sip:10.31.255.134:5070 SIP/2.0
00009 1732188190 * <== 10.31.255.134:5070       SIP/2.0 200 OK
00010 1732188192 * <== 192.168.0.1:58892        INVITE sip:111@ha.MY_DOMAIN.com SIP/2.0
00011 1732188192 * ==> 192.168.0.1:58892        SIP/2.0 401 Unauthorized
00012 1732188192 * <== 192.168.0.1:58892        ACK sip:111@ha.MY_DOMAIN.com SIP/2.0
00013 1732188192 * <== 192.168.0.1:58892        INVITE sip:111@ha.MY_DOMAIN.com SIP/2.0
00014 1732188192 * ==> 192.168.0.1:58892        SIP/2.0 100 Trying
00015 1732188192 * ==> 192.168.0.222:5060       INVITE sip:111@192.168.0.222:5060 SIP/2.0
00016 1732188192 * <== 192.168.0.222:5060       SIP/2.0 100 Trying
00017 1732188193 * <== 192.168.0.222:5060       SIP/2.0 404 Not Found
00018 1732188193 * ==> 192.168.0.222:5060       ACK sip:111@192.168.0.222:5060 SIP/2.0
00019 1732188193 * ==> 192.168.0.1:58892        SIP/2.0 503 Service Unavailable
00020 1732188193 * <== 192.168.0.1:58892        ACK sip:111@ha.MY_DOMAIN.com SIP/2.0

jcolp · November 21, 2024, 11:34am

Those are not the complete messages. It just confirms that whatever is at 192.168.0.222 responded with the 404. Why that is, no idea. Having complete messages can sometimes show the difference and provide a further hint as to what.

sesardelaisla · November 21, 2024, 11:37am

I got such messages with the pjsip set history on and pjsip show history commands through Asterisk CLI. Any advice on how to get complete messages, please? Would getting all sip messages with Wireshark help for the purpose or is there any cleaner way to achieve this?

jcolp · November 21, 2024, 11:41am

The “pjsip set logger on” CLI command will output SIP traffic to the Asterisk console. A packet capture would also capture the SIP traffic.

My gut says, though, that ultimately it won’t show anything really different and it’ll be guessing of what to do (if anything) to appease the older device - such as tweaking configuration and seeing what happens, or perhaps configuration being different between the two devices in some subtle way resulting in different behavior.

sesardelaisla · November 21, 2024, 12:22pm

.222 is the bad doorbell. .72 is the good one.

I hope this is the right logging thing. It is what I get on the console after issuing the pjsip set logger on:

BAD DOORBELL (Registration and call to doorbell - 111)

    -- Added contact 'sip:111@192.168.0.222:5060' to AOR '111' with expiration of 3600 seconds
  == Endpoint 111 is now Reachable
    -- Executing [111@default:1] Dial("PJSIP/webrtc_client-0000000a", "PJSIP/111") in new stack
    -- Called PJSIP/111
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'PJSIP/webrtc_client-0000000a' status is 'CHANUNAVAIL'

GOOD DOORBELL (Registration and call to doorbell - 111)

    -- Added contact 'sip:111@192.168.0.72:5060' to AOR '111' with expiration of 3600 seconds
  == Endpoint 111 is now Reachable
    -- Executing [111@default:1] Dial("PJSIP/webrtc_client-0000000c", "PJSIP/111") in new stack
    -- Called PJSIP/111
       > 0x7f7750044900 -- Strict RTP learning after remote address set to: 192.168.0.72:9654
    -- PJSIP/111-0000000d answered PJSIP/webrtc_client-0000000c
       > 0x7f779c023730 -- Strict RTP learning after remote address set to: 80.x.x.x:50741
    -- Channel PJSIP/111-0000000d joined 'simple_bridge' basic-bridge <c53108be-c6ac-48de-8f76-a6928c92916b>
    -- Channel PJSIP/webrtc_client-0000000c joined 'simple_bridge' basic-bridge <c53108be-c6ac-48de-8f76-a6928c92916b>
       > 0x7f779c023730 -- Strict RTP learning after ICE completion
       > 0x7f7750044900 -- Strict RTP switching to RTP target address 192.168.0.72:9654 as source
       > 0x7f779c023730 -- Strict RTP learning after remote address set to: 192.168.0.41:50741
       > 0x7f779c023730 -- Strict RTP switching to RTP target address 192.168.0.41:50741 as source
       > 0x7f779c023730 -- Strict RTP learning complete - Locking on source address 192.168.0.41:50741
       > 0x7f7750044900 -- Strict RTP learning complete - Locking on source address 192.168.0.72:9654
    -- Channel PJSIP/webrtc_client-0000000c left 'simple_bridge' basic-bridge <c53108be-c6ac-48de-8f76-a6928c92916b>
    -- Channel PJSIP/111-0000000d left 'simple_bridge' basic-bridge <c53108be-c6ac-48de-8f76-a6928c92916b>
  == Spawn extension (default, 111, 1) exited non-zero on 'PJSIP/webrtc_client-0000000c'

I understand that it will be challenging to find the actual problem without being sure how the bad doorbell/firmware manages SIP. I restored it to factory and configured it from scratch to see whether it made a difference. I set everything in the same way the good doorbell is configured. There are not many settings to set so I am pretty sure it is not a config issue, but some sort of issue related to the fact that the old one is discontinued. However, I would like to try whatever I can before surrendering. Any advice will be much appreciated, but if it becomes not possible to use the SIP thing for my purpose, I will just use such doorbell for other purpose. Just keen on knowing what it is really happening. Happy to get even more logs with Wireshark if you think it might help. In the meantime, I will double check all settings, try different firwares and ohter stuff. Thanks.

jcolp · November 21, 2024, 12:23pm

There’s no SIP traffic in there. Did you do anything else after issuing “pjsip set logger on” such as reload?

sesardelaisla · November 21, 2024, 12:38pm

Nop. But I just restarted Asterisk and issued the command again. Output is different now and I guess it is what you were expecting at last:

[Splitting message due to character count limit]

GOOD DOORBELL (Registration and call to doorbell - 111)

<--- Received SIP request (443 bytes) from UDP:192.168.0.72:5060 --->
REGISTER sip:192.168.0.100:5060 SIP/2.0
Contact: "Doorbell" <sip:111@192.168.0.72:5060>
Expires: 3600
To: <sip:111@192.168.0.100:5060>
Via: SIP/2.0/UDP 192.168.0.72:5060;rport;branch=z9hG4bK702631525
From: "Doorbell" <sip:111@192.168.0.100:5060>;tag=553852927
Call-ID: 965155836@192.168.0.100:5060
CSeq: 8 REGISTER
User-Agent: YATE/5.5.0
Max-Forwards: 70
Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, OPTIONS, INFO
Content-Length: 0


<--- Transmitting SIP response (479 bytes) to UDP:192.168.0.72:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.72:5060;rport=5060;received=192.168.0.72;branch=z9hG4bK702631525
Call-ID: 965155836@192.168.0.100:5060
From: "Doorbell" <sip:111@192.168.0.100>;tag=553852927
To: <sip:111@192.168.0.100>;tag=z9hG4bK702631525
CSeq: 8 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1732192078/1d12be88c1d318d98e65830c930fc3e4",opaque="5c0e3e4f29d4d997",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.8.1
Content-Length:  0


<--- Received SIP request (736 bytes) from UDP:192.168.0.72:5060 --->
REGISTER sip:192.168.0.100:5060 SIP/2.0
Contact: "Doorbell" <sip:111@192.168.0.72:5060>
Expires: 3600
To: <sip:111@192.168.0.100:5060>
Via: SIP/2.0/UDP 192.168.0.72:5060;rport;branch=z9hG4bK2097520546
From: "Doorbell" <sip:111@192.168.0.100:5060>;tag=553852927
Call-ID: 965155836@192.168.0.100:5060
User-Agent: YATE/5.5.0
Max-Forwards: 70
Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, OPTIONS, INFO
CSeq: 9 REGISTER
Authorization: Digest username="111", realm="asterisk", nonce="1732192078/1d12be88c1d318d98e65830c930fc3e4", uri="sip:192.168.0.100:5060", response="d3192c0f5476a660870d788ae600ec96", algorithm=MD5, opaque="5c0e3e4f29d4d997", qop=auth, nc=00000015, cnonce="1e518cf3728a46fc9f29656fcfb73323"
Content-Length: 0


    -- Added contact 'sip:111@192.168.0.72:5060' to AOR '111' with expiration of 3600 seconds
<--- Transmitting SIP response (428 bytes) to UDP:192.168.0.72:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.72:5060;rport=5060;received=192.168.0.72;branch=z9hG4bK2097520546
Call-ID: 965155836@192.168.0.100:5060
From: "Doorbell" <sip:111@192.168.0.100>;tag=553852927
To: <sip:111@192.168.0.100>;tag=z9hG4bK2097520546
CSeq: 9 REGISTER
Date: Thu, 21 Nov 2024 12:27:58 GMT
Contact: <sip:111@192.168.0.72:5060>;expires=3599
Expires: 3600
Server: Asterisk PBX 20.8.1
Content-Length:  0


  == Endpoint 111 is now Reachable
<--- Received SIP request (2029 bytes) from WSS:192.168.0.1:59163 --->
INVITE sip:111@ha.balimelon.com SIP/2.0
Via: SIP/2.0/WSS 9m5rnafgr7ue.invalid;branch=z9hG4bK3170840
Max-Forwards: 69
To: <sip:111@ha.balimelon.com>
From: <sip:webrtc_client@ha.balimelon.com>;tag=kq12b6aq7n
Call-ID: 18oi8tn3bp6avkov6df5
CSeq: 7213 INVITE
Contact: <sip:e4d5a1op@9m5rnafgr7ue.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: timer,ice,replaces,outbound
User-Agent: JsSIP 3.9.0
Content-Length: 1494

v=0
o=mozilla...THIS_IS_SDPARTA-99.0 5439934695468138311 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 EE:0D:AE:FF:CC:06:67:03:19:DE:EB:42:1E:75:AA:52:AD:AA:3D:FD:FF:5F:5C:65:C0:01:77:D4:96:45:75:89
a=group:BUNDLE 0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 51280 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 80.26.218.23
a=candidate:0 1 UDP 2122252543 192.168.0.41 51280 typ host
a=candidate:2 1 TCP 2105524479 192.168.0.41 9 typ host tcptype active
a=candidate:0 2 UDP 2122252542 192.168.0.41 53639 typ host
a=candidate:2 2 TCP 2105524478 192.168.0.41 9 typ host tcptype active
a=candidate:1 1 UDP 1686052863 80.26.218.23 51280 typ srflx raddr 192.168.0.41 rport 51280
a=candidate:1 2 UDP 1686052862 80.26.218.23 53639 typ srflx raddr 192.168.0.41 rport 53639
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:6e2e8e3624f641427eac69cc3325a987
a=ice-ufrag:d637ba95
a=mid:0
a=msid:{a4caafa9-eb72-46e7-8972-88460c672908} {ac79493b-22da-4e53-8461-0fbb886beb19}
a=rtcp:53639 IN IP4 80.26.218.23
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=setup:actpass
a=ssrc:4267818680 cname:{604316b0-86ae-4a34-9d64-797ab679f8b8}

<--- Transmitting SIP response (477 bytes) to WSS:192.168.0.1:59163 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 9m5rnafgr7ue.invalid;rport=59163;received=192.168.0.1;branch=z9hG4bK3170840
Call-ID: 18oi8tn3bp6avkov6df5
From: <sip:webrtc_client@ha.balimelon.com>;tag=kq12b6aq7n
To: <sip:111@ha.balimelon.com>;tag=z9hG4bK3170840
CSeq: 7213 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1732192087/fb150d4363a481934a3b2b34e1f2cd52",opaque="796d6fe631cfe23f",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.8.1
Content-Length:  0


<--- Received SIP request (412 bytes) from WSS:192.168.0.1:59163 --->
ACK sip:111@ha.balimelon.com SIP/2.0
Via: SIP/2.0/WSS 9m5rnafgr7ue.invalid;branch=z9hG4bK3170840
Max-Forwards: 69
To: <sip:111@ha.balimelon.com>;tag=z9hG4bK3170840
From: <sip:webrtc_client@ha.balimelon.com>;tag=kq12b6aq7n
Call-ID: 18oi8tn3bp6avkov6df5
CSeq: 7213 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: JsSIP 3.9.0
Content-Length: 0


<--- Received SIP request (2313 bytes) from WSS:192.168.0.1:59163 --->
INVITE sip:111@ha.balimelon.com SIP/2.0
Via: SIP/2.0/WSS 9m5rnafgr7ue.invalid;branch=z9hG4bK8309065
Max-Forwards: 69
To: <sip:111@ha.balimelon.com>
From: <sip:webrtc_client@ha.balimelon.com>;tag=kq12b6aq7n
Call-ID: 18oi8tn3bp6avkov6df5
CSeq: 7214 INVITE
Authorization: Digest algorithm=MD5, username="webrtc_client", realm="asterisk", nonce="1732192087/fb150d4363a481934a3b2b34e1f2cd52", uri="sip:111@ha.balimelon.com", response="b75cc38de212798399c131644774eee8", opaque="796d6fe631cfe23f", qop=auth, cnonce="0dvncgedq7b6", nc=00000001
Contact: <sip:e4d5a1op@9m5rnafgr7ue.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: timer,ice,replaces,outbound
User-Agent: JsSIP 3.9.0
Content-Length: 1494

v=0
o=mozilla...THIS_IS_SDPARTA-99.0 5439934695468138311 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 EE:0D:AE:FF:CC:06:67:03:19:DE:EB:42:1E:75:AA:52:AD:AA:3D:FD:FF:5F:5C:65:C0:01:77:D4:96:45:75:89
a=group:BUNDLE 0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 51280 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 80.26.218.23
a=candidate:0 1 UDP 2122252543 192.168.0.41 51280 typ host
a=candidate:2 1 TCP 2105524479 192.168.0.41 9 typ host tcptype active
a=candidate:0 2 UDP 2122252542 192.168.0.41 53639 typ host
a=candidate:2 2 TCP 2105524478 192.168.0.41 9 typ host tcptype active
a=candidate:1 1 UDP 1686052863 80.26.218.23 51280 typ srflx raddr 192.168.0.41 rport 51280
a=candidate:1 2 UDP 1686052862 80.26.218.23 53639 typ srflx raddr 192.168.0.41 rport 53639
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:6e2e8e3624f641427eac69cc3325a987
a=ice-ufrag:d637ba95
a=mid:0
a=msid:{a4caafa9-eb72-46e7-8972-88460c672908} {ac79493b-22da-4e53-8461-0fbb886beb19}
a=rtcp:53639 IN IP4 80.26.218.23
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=setup:actpass
a=ssrc:4267818680 cname:{604316b0-86ae-4a34-9d64-797ab679f8b8}

<--- Transmitting SIP response (306 bytes) to WSS:192.168.0.1:59163 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS 9m5rnafgr7ue.invalid;rport=59163;received=192.168.0.1;branch=z9hG4bK8309065
Call-ID: 18oi8tn3bp6avkov6df5
From: <sip:webrtc_client@ha.balimelon.com>;tag=kq12b6aq7n
To: <sip:111@ha.balimelon.com>
CSeq: 7214 INVITE
Server: Asterisk PBX 20.8.1
Content-Length:  0


    -- Executing [111@default:1] Dial("PJSIP/webrtc_client-00000001", "PJSIP/111") in new stack
    -- Called PJSIP/111
<--- Transmitting SIP request (890 bytes) to UDP:192.168.0.72:5060 --->
INVITE sip:111@192.168.0.72:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.100:5060;rport;branch=z9hG4bKPja3226738-3113-4c68-87a9-dba358d2c747
From: "HASS" <sip:webrtc_client@192.168.0.100>;tag=5b8ad7ff-db4a-4180-9937-b5bf8dd158c1
To: <sip:111@192.168.0.72>
Contact: <sip:asterisk@192.168.0.100:5060>
Call-ID: f7a1217a-e462-4362-b86e-a40e87891df3
CSeq: 10802 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, INFO
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.8.1
Content-Type: application/sdp
Content-Length:   207

v=0
o=- 1321765909 1321765909 IN IP4 192.168.0.100
s=Asterisk
c=IN IP4 192.168.0.100
t=0 0
m=audio 19458 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Received SIP response (369 bytes) from UDP:192.168.0.72:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.100:5060;rport=5060;branch=z9hG4bKPja3226738-3113-4c68-87a9-dba358d2c747;received=192.168.0.100
From: "HASS" <sip:webrtc_client@192.168.0.100>;tag=5b8ad7ff-db4a-4180-9937-b5bf8dd158c1
To: <sip:111@192.168.0.72>
Call-ID: f7a1217a-e462-4362-b86e-a40e87891df3
CSeq: 10802 INVITE
Server: YATE/5.5.0
Content-Length: 0


<--- Received SIP response (770 bytes) from UDP:192.168.0.72:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.100:5060;rport=5060;branch=z9hG4bKPja3226738-3113-4c68-87a9-dba358d2c747;received=192.168.0.100
From: "HASS" <sip:webrtc_client@192.168.0.100>;tag=5b8ad7ff-db4a-4180-9937-b5bf8dd158c1
To: <sip:111@192.168.0.72>;tag=686487446
Call-ID: f7a1217a-e462-4362-b86e-a40e87891df3
CSeq: 10802 INVITE
Server: YATE/5.5.0
Contact: <sip:111@192.168.0.72:5060>
Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, OPTIONS, INFO
Content-Type: application/sdp
Content-Length: 263

v=0
o=AZ5067601 0 0 IN IP4 192.168.0.72
s=Talk session
c=IN IP4 192.168.0.72
t=0 0
m=audio 9654 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=sendrecv
m=video 9856 RTP/AVP 96
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42801F;packetization-mode=1
a=sendonly

       > 0x7f52e00d1a30 -- Strict RTP learning after remote address set to: 192.168.0.72:9654
<--- Transmitting SIP request (403 bytes) to UDP:192.168.0.72:5060 --->
ACK sip:111@192.168.0.72:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.100:5060;rport;branch=z9hG4bKPj2a84d2ec-b7b7-4706-8a12-00debd0062fe
From: "HASS" <sip:webrtc_client@192.168.0.100>;tag=5b8ad7ff-db4a-4180-9937-b5bf8dd158c1
To: <sip:111@192.168.0.72>;tag=686487446
Call-ID: f7a1217a-e462-4362-b86e-a40e87891df3
CSeq: 10802 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 20.8.1
Content-Length:  0


    -- PJSIP/111-00000002 answered PJSIP/webrtc_client-00000001
       > 0x7f52e0117890 -- Strict RTP learning after remote address set to: 80.26.218.23:51280
<--- Transmitting SIP response (4227 bytes) to WSS:192.168.0.1:59163 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 9m5rnafgr7ue.invalid;rport=59163;received=192.168.0.1;branch=z9hG4bK8309065
Call-ID: 18oi8tn3bp6avkov6df5
From: <sip:webrtc_client@ha.balimelon.com>;tag=kq12b6aq7n
To: <sip:111@ha.balimelon.com>;tag=7e07c8ce-98cc-468c-8e97-9240371b52b6
CSeq: 7214 INVITE
Server: Asterisk PBX 20.8.1
Contact: <sip:192.168.0.100:18089;transport=ws>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, INFO
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 90;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:  3585

v=0
o=- 5439934695468138311 2 IN IP4 192.168.0.100
s=Asterisk
c=IN IP4 192.168.0.100
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0
m=audio 12016 UDP/TLS/RTP/SAVPF 109 8 0 101
a=connection:new
a=setup:active
a=fingerprint:SHA-256 FF:53:9A:20:98:73:66:8B:52:6A:AA:B4:FA:E4:1E:8F:F6:7D:88:61:FC:F3:89:A3:AE:E4:72:40:44:26:8A:6B
a=ice-ufrag:60dd4a6f4ec28c241457c9641e134220
a=ice-pwd:6353541c0c4fe4613f960ff932741d48
a=candidate:Hc0a80064 1 UDP 2130706431 192.168.0.100 12016 typ host
a=candidate:Hac1e2001 1 UDP 2130706431 172.30.32.1 12016 typ host
a=candidate:Hac1ee801 1 UDP 2130706431 172.30.232.1 12016 typ host
a=candidate:H953aa113 1 UDP 2130706431 fdb2:65d3:8262:ce4c:507:b6fd:82e7:fbcb 12016 typ host
a=candidate:H725bf929 1 UDP 2130706431 fe80::3554:32f6:41c4:f362 12016 typ host
a=candidate:H96e485e 1 UDP 2130706431 fe80::42:96ff:fe23:a206 12016 typ host
a=candidate:H35c0444d 1 UDP 2130706431 fe80::42:a9ff:fe24:3dc6 12016 typ host
a=candidate:Hcb447602 1 UDP 2130706431 fe80::c8c8:1cff:fe7b:5e02 12016 typ host
a=candidate:H6ce55853 1 UDP 2130706431 fe80::6005:37ff:fe8d:cec1 12016 typ host
a=candidate:H905c4b9c 1 UDP 2130706431 fe80::14b7:1cff:fe5e:9b21 12016 typ host
a=candidate:H9c4c3bf0 1 UDP 2130706431 fe80::64a6:40ff:fec6:a4c1 12016 typ host
a=candidate:H684f9b23 1 UDP 2130706431 fe80::8858:43ff:fee6:2c93 12016 typ host
a=candidate:H7ffc8c69 1 UDP 2130706431 fe80::2c37:a8ff:fe7b:5deb 12016 typ host
a=candidate:Hb01fbdca 1 UDP 2130706431 fe80::5cc4:deff:fec6:5ead 12016 typ host
a=candidate:H32007257 1 UDP 2130706431 fe80::38b2:80ff:feba:9bbd 12016 typ host
a=candidate:H107dc5e 1 UDP 2130706431 fe80::185e:54ff:fef6:3310 12016 typ host
a=candidate:H931c18b1 1 UDP 2130706431 fe80::1475:7aff:fe06:a964 12016 typ host
a=candidate:Hc4f51623 1 UDP 2130706431 fe80::486f:7bff:fe80:8e88 12016 typ host
a=candidate:Heb9094a4 1 UDP 2130706431 fe80::2ca7:1eff:fecd:4cdf 12016 typ host
a=candidate:H8c22c91e 1 UDP 2130706431 fe80::909e:23ff:fe2e:90b4 12016 typ host
a=candidate:H2313eab3 1 UDP 2130706431 fe80::e45e:b3ff:fe17:8d5f 12016 typ host
a=candidate:H481ad450 1 UDP 2130706431 fe80::aa:93ff:fef0:1cac 12016 typ host
a=candidate:H6d1a37a2 1 UDP 2130706431 fe80::2cc6:a0ff:fe3d:a078 12016 typ host
a=candidate:Hd5eabb6b 1 UDP 2130706431 fe80::346d:1bff:fe04:ef41 12016 typ host
a=candidate:Hb00feead 1 UDP 2130706431 fe80::ece7:b0ff:fea3:8ac2 12016 typ host
a=candidate:Ha4af36ef 1 UDP 2130706431 fd6c:4a29:58d9:23b5:0:ff:fe00:fc11 12016 typ host
a=candidate:H7452e55f 1 UDP 2130706431 fdf9:8954:8f0b:1:d42:2e72:3f43:1a06 12016 typ host
a=candidate:Ha4af36ee 1 UDP 2130706431 fd6c:4a29:58d9:23b5:0:ff:fe00:fc10 12016 typ host
a=candidate:Ha4af3716 1 UDP 2130706431 fd6c:4a29:58d9:23b5:0:ff:fe00:fc38 12016 typ host
a=candidate:Ha4af36de 1 UDP 2130706431 fd6c:4a29:58d9:23b5:0:ff:fe00:fc00 12016 typ host
a=candidate:Ha4af1c0e 1 UDP 2130706431 fd6c:4a29:58d9:23b5:0:ff:fe00:2c00 12016 typ host
a=candidate:Hed520d50 1 UDP 2130706431 fd6c:4a29:58d9:23b5:edf6:b372:54e5:e1c 12016 typ host
a=candidate:Hbf78e1a1 1 UDP 2130706431 fe80::d0f5:22cf:3065:c9ef 12016 typ host
a=candidate:S501ada17 1 UDP 1694498815 80.26.218.23 12016 typ srflx raddr 192.168.0.100 rport 12016
a=rtpmap:109 opus/48000/2
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
a=rtcp-mux
a=ssrc:824679544 cname:6f750936-a6a1-4808-8e9e-7b1a62420948
a=msid:a058ec77-deda-4c7f-96f0-d31454c76da2 9a33a347-3f89-4ce6-9a79-9a23d69f3033
a=rtcp-fb:* transport-cc
a=mid:0

    -- Channel PJSIP/111-00000002 joined 'simple_bridge' basic-bridge <3bacf078-3be6-4388-853e-390383629fd8>
    -- Channel PJSIP/webrtc_client-00000001 joined 'simple_bridge' basic-bridge <3bacf078-3be6-4388-853e-390383629fd8>
       > 0x7f52e00d1a30 -- Strict RTP switching to RTP target address 192.168.0.72:9654 as source
       > 0x7f52e0117890 -- Strict RTP learning after ICE completion
<--- Received SIP request (446 bytes) from WSS:192.168.0.1:59163 --->
ACK sip:192.168.0.100:18089;transport=ws SIP/2.0
Via: SIP/2.0/WSS 9m5rnafgr7ue.invalid;branch=z9hG4bK5349959
Max-Forwards: 69
To: <sip:111@ha.balimelon.com>;tag=7e07c8ce-98cc-468c-8e97-9240371b52b6
From: <sip:webrtc_client@ha.balimelon.com>;tag=kq12b6aq7n
Call-ID: 18oi8tn3bp6avkov6df5
CSeq: 7214 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: JsSIP 3.9.0
Content-Length: 0


       > 0x7f52e0117890 -- Strict RTP learning after remote address set to: 192.168.0.41:51280
       > 0x7f52e0117890 -- Strict RTP switching to RTP target address 192.168.0.41:51280 as source
<--- Received SIP request (446 bytes) from WSS:192.168.0.1:59163 --->
BYE sip:192.168.0.100:18089;transport=ws SIP/2.0
Via: SIP/2.0/WSS 9m5rnafgr7ue.invalid;branch=z9hG4bK6616249
Max-Forwards: 69
To: <sip:111@ha.balimelon.com>;tag=7e07c8ce-98cc-468c-8e97-9240371b52b6
From: <sip:webrtc_client@ha.balimelon.com>;tag=kq12b6aq7n
Call-ID: 18oi8tn3bp6avkov6df5
CSeq: 7215 BYE
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: JsSIP 3.9.0
Content-Length: 0


<--- Transmitting SIP response (340 bytes) to WSS:192.168.0.1:59163 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 9m5rnafgr7ue.invalid;rport=59163;received=192.168.0.1;branch=z9hG4bK6616249
Call-ID: 18oi8tn3bp6avkov6df5
From: <sip:webrtc_client@ha.balimelon.com>;tag=kq12b6aq7n
To: <sip:111@ha.balimelon.com>;tag=7e07c8ce-98cc-468c-8e97-9240371b52b6
CSeq: 7215 BYE
Server: Asterisk PBX 20.8.1
Content-Length:  0


    -- Channel PJSIP/webrtc_client-00000001 left 'simple_bridge' basic-bridge <3bacf078-3be6-4388-853e-390383629fd8>
  == Spawn extension (default, 111, 1) exited non-zero on 'PJSIP/webrtc_client-00000001'
    -- Channel PJSIP/111-00000002 left 'simple_bridge' basic-bridge <3bacf078-3be6-4388-853e-390383629fd8>
<--- Transmitting SIP request (427 bytes) to UDP:192.168.0.72:5060 --->
BYE sip:111@192.168.0.72:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.100:5060;rport;branch=z9hG4bKPj160e6756-d77c-4217-bfde-66240e45ba2e
From: "HASS" <sip:webrtc_client@192.168.0.100>;tag=5b8ad7ff-db4a-4180-9937-b5bf8dd158c1
To: <sip:111@192.168.0.72>;tag=686487446
Call-ID: f7a1217a-e462-4362-b86e-a40e87891df3
CSeq: 10803 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 20.8.1
Content-Length:  0


<--- Received SIP response (380 bytes) from UDP:192.168.0.72:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.100:5060;rport=5060;branch=z9hG4bKPj160e6756-d77c-4217-bfde-66240e45ba2e;received=192.168.0.100
From: "HASS" <sip:webrtc_client@192.168.0.100>;tag=5b8ad7ff-db4a-4180-9937-b5bf8dd158c1
To: <sip:111@192.168.0.72>;tag=686487446
Call-ID: f7a1217a-e462-4362-b86e-a40e87891df3
CSeq: 10803 BYE
Server: YATE/5.5.0
Content-Length: 0


<--- Received SIP response (477 bytes) from UDP:192.168.0.72:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.100:5060;rport=5060;branch=z9hG4bKPj160e6756-d77c-4217-bfde-66240e45ba2e;received=192.168.0.100
From: "HASS" <sip:webrtc_client@192.168.0.100>;tag=5b8ad7ff-db4a-4180-9937-b5bf8dd158c1
To: <sip:111@192.168.0.72>;tag=686487446
Call-ID: f7a1217a-e462-4362-b86e-a40e87891df3
CSeq: 10803 BYE
P-RTP-Stat: PS=0,OS=0,PR=187,OR=29920,PL=0
Server: YATE/5.5.0
Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, OPTIONS, INFO
Content-Length: 0

sesardelaisla · November 21, 2024, 12:38pm

BAD DOORBELL (Registration and call to doorbell - 111)

<--- Received SIP request (444 bytes) from UDP:192.168.0.222:5060 --->
REGISTER sip:192.168.0.100:5060 SIP/2.0
Contact: "Doorbell" <sip:111@192.168.0.222:5060>
Expires: 0
To: <sip:111@192.168.0.100:5060>
Call-ID: 1104591129@192.168.0.100:5060
Via: SIP/2.0/UDP 192.168.0.222:5060;rport;branch=z9hG4bK634357201
From: "Doorbell" <sip:111@192.168.0.100:5060>;tag=1955946774
CSeq: 4 REGISTER
User-Agent: YATE/5.5.0
Max-Forwards: 70
Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, OPTIONS, INFO
Content-Length: 0


<--- Transmitting SIP response (483 bytes) to UDP:192.168.0.222:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.222:5060;rport=5060;received=192.168.0.222;branch=z9hG4bK634357201
Call-ID: 1104591129@192.168.0.100:5060
From: "Doorbell" <sip:111@192.168.0.100>;tag=1955946774
To: <sip:111@192.168.0.100>;tag=z9hG4bK634357201
CSeq: 4 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1732192263/1f4e53f7a36222a1ea1d8b52df67270d",opaque="60d89df86d8fc492",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.8.1
Content-Length:  0


<--- Received SIP request (737 bytes) from UDP:192.168.0.222:5060 --->
REGISTER sip:192.168.0.100:5060 SIP/2.0
Contact: "Doorbell" <sip:111@192.168.0.222:5060>
Expires: 0
To: <sip:111@192.168.0.100:5060>
Call-ID: 1104591129@192.168.0.100:5060
Via: SIP/2.0/UDP 192.168.0.222:5060;rport;branch=z9hG4bK1199075116
From: "Doorbell" <sip:111@192.168.0.100:5060>;tag=1955946774
User-Agent: YATE/5.5.0
Max-Forwards: 70
Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, OPTIONS, INFO
CSeq: 5 REGISTER
Authorization: Digest username="111", realm="asterisk", nonce="1732192263/1f4e53f7a36222a1ea1d8b52df67270d", uri="sip:192.168.0.100:5060", response="ed11c8abdde521766577121c7ac59640", algorithm=MD5, opaque="60d89df86d8fc492", qop=auth, nc=00000004, cnonce="32068d16d522f7376c84f228b2a5838f"
Content-Length: 0


    -- Attempted to remove non-existent contact 'sip:111@192.168.0.222:5060' from AOR '111' by request
<--- Transmitting SIP response (378 bytes) to UDP:192.168.0.222:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.222:5060;rport=5060;received=192.168.0.222;branch=z9hG4bK1199075116
Call-ID: 1104591129@192.168.0.100:5060
From: "Doorbell" <sip:111@192.168.0.100>;tag=1955946774
To: <sip:111@192.168.0.100>;tag=z9hG4bK1199075116
CSeq: 5 REGISTER
Date: Thu, 21 Nov 2024 12:31:03 GMT
Expires: 0
Server: Asterisk PBX 20.8.1
Content-Length:  0


<--- Received SIP request (447 bytes) from UDP:192.168.0.222:5060 --->
REGISTER sip:192.168.0.100:5060 SIP/2.0
Contact: "Doorbell" <sip:111@192.168.0.222:5060>
Expires: 3600
To: <sip:111@192.168.0.100:5060>
Via: SIP/2.0/UDP 192.168.0.222:5060;rport;branch=z9hG4bK1477237725
From: "Doorbell" <sip:111@192.168.0.100:5060>;tag=545668946
Call-ID: 1708491931@192.168.0.100:5060
CSeq: 3 REGISTER
User-Agent: YATE/5.5.0
Max-Forwards: 70
Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, OPTIONS, INFO
Content-Length: 0


<--- Transmitting SIP response (484 bytes) to UDP:192.168.0.222:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.222:5060;rport=5060;received=192.168.0.222;branch=z9hG4bK1477237725
Call-ID: 1708491931@192.168.0.100:5060
From: "Doorbell" <sip:111@192.168.0.100>;tag=545668946
To: <sip:111@192.168.0.100>;tag=z9hG4bK1477237725
CSeq: 3 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1732192264/41df43a0b1fda53f6b284738282492ea",opaque="1a7599a2732a0072",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.8.1
Content-Length:  0


<--- Received SIP request (737 bytes) from UDP:192.168.0.222:5060 --->
REGISTER sip:192.168.0.100:5060 SIP/2.0
Contact: "Doorbell" <sip:111@192.168.0.222:5060>
Expires: 3600
To: <sip:111@192.168.0.100:5060>
Via: SIP/2.0/UDP 192.168.0.222:5060;rport;branch=z9hG4bK63024937
From: "Doorbell" <sip:111@192.168.0.100:5060>;tag=545668946
Call-ID: 1708491931@192.168.0.100:5060
User-Agent: YATE/5.5.0
Max-Forwards: 70
Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, OPTIONS, INFO
CSeq: 4 REGISTER
Authorization: Digest username="111", realm="asterisk", nonce="1732192264/41df43a0b1fda53f6b284738282492ea", uri="sip:192.168.0.100:5060", response="52fe545ca29e8ae057dba06bbfc64bc0", algorithm=MD5, opaque="1a7599a2732a0072", qop=auth, nc=00000005, cnonce="804b67f511b08d21557a45bd0f1cf803"
Content-Length: 0


    -- Added contact 'sip:111@192.168.0.222:5060' to AOR '111' with expiration of 3600 seconds
<--- Transmitting SIP response (428 bytes) to UDP:192.168.0.222:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.222:5060;rport=5060;received=192.168.0.222;branch=z9hG4bK63024937
Call-ID: 1708491931@192.168.0.100:5060
From: "Doorbell" <sip:111@192.168.0.100>;tag=545668946
To: <sip:111@192.168.0.100>;tag=z9hG4bK63024937
CSeq: 4 REGISTER
Date: Thu, 21 Nov 2024 12:31:04 GMT
Contact: <sip:111@192.168.0.222:5060>;expires=3599
Expires: 3600
Server: Asterisk PBX 20.8.1
Content-Length:  0


  == Endpoint 111 is now Reachable
<--- Received SIP request (2029 bytes) from WSS:192.168.0.1:59163 --->
INVITE sip:111@ha.balimelon.com SIP/2.0
Via: SIP/2.0/WSS 9m5rnafgr7ue.invalid;branch=z9hG4bK8028580
Max-Forwards: 69
To: <sip:111@ha.balimelon.com>
From: <sip:webrtc_client@ha.balimelon.com>;tag=q2md5opiei
Call-ID: 18oi83ttq3s21kl31c6a
CSeq: 6233 INVITE
Contact: <sip:e4d5a1op@9m5rnafgr7ue.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: timer,ice,replaces,outbound
User-Agent: JsSIP 3.9.0
Content-Length: 1494

v=0
o=mozilla...THIS_IS_SDPARTA-99.0 4935802565320084750 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 7E:AB:AC:A0:5B:7C:F6:6A:62:95:DC:ED:C3:D2:28:EA:44:97:26:A6:1B:13:D0:EF:66:C4:E3:CE:AB:4B:D6:36
a=group:BUNDLE 0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 63803 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 80.26.218.23
a=candidate:0 1 UDP 2122252543 192.168.0.41 63803 typ host
a=candidate:2 1 TCP 2105524479 192.168.0.41 9 typ host tcptype active
a=candidate:0 2 UDP 2122252542 192.168.0.41 63304 typ host
a=candidate:2 2 TCP 2105524478 192.168.0.41 9 typ host tcptype active
a=candidate:1 1 UDP 1686052863 80.26.218.23 63803 typ srflx raddr 192.168.0.41 rport 63803
a=candidate:1 2 UDP 1686052862 80.26.218.23 63304 typ srflx raddr 192.168.0.41 rport 63304
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:5c72374c75e7beae1681991bf9be2189
a=ice-ufrag:35dae94a
a=mid:0
a=msid:{6172abf6-4d47-411d-bc67-721b581b5a77} {0f0506fb-5c0e-41a1-ba59-96218d305cf7}
a=rtcp:63304 IN IP4 80.26.218.23
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=setup:actpass
a=ssrc:2585001976 cname:{4784dd3a-3bb1-4b62-9897-cc649f35f743}

<--- Transmitting SIP response (477 bytes) to WSS:192.168.0.1:59163 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 9m5rnafgr7ue.invalid;rport=59163;received=192.168.0.1;branch=z9hG4bK8028580
Call-ID: 18oi83ttq3s21kl31c6a
From: <sip:webrtc_client@ha.balimelon.com>;tag=q2md5opiei
To: <sip:111@ha.balimelon.com>;tag=z9hG4bK8028580
CSeq: 6233 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1732192267/ab151cd9a7d7ad98550060ff4a2185d1",opaque="1059561c28d4b502",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.8.1
Content-Length:  0


<--- Received SIP request (412 bytes) from WSS:192.168.0.1:59163 --->
ACK sip:111@ha.balimelon.com SIP/2.0
Via: SIP/2.0/WSS 9m5rnafgr7ue.invalid;branch=z9hG4bK8028580
Max-Forwards: 69
To: <sip:111@ha.balimelon.com>;tag=z9hG4bK8028580
From: <sip:webrtc_client@ha.balimelon.com>;tag=q2md5opiei
Call-ID: 18oi83ttq3s21kl31c6a
CSeq: 6233 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: JsSIP 3.9.0
Content-Length: 0


<--- Received SIP request (2313 bytes) from WSS:192.168.0.1:59163 --->
INVITE sip:111@ha.balimelon.com SIP/2.0
Via: SIP/2.0/WSS 9m5rnafgr7ue.invalid;branch=z9hG4bK1807946
Max-Forwards: 69
To: <sip:111@ha.balimelon.com>
From: <sip:webrtc_client@ha.balimelon.com>;tag=q2md5opiei
Call-ID: 18oi83ttq3s21kl31c6a
CSeq: 6234 INVITE
Authorization: Digest algorithm=MD5, username="webrtc_client", realm="asterisk", nonce="1732192267/ab151cd9a7d7ad98550060ff4a2185d1", uri="sip:111@ha.balimelon.com", response="89af6ee31ed91489681a63e30586f272", opaque="1059561c28d4b502", qop=auth, cnonce="q46ra69jrjti", nc=00000001
Contact: <sip:e4d5a1op@9m5rnafgr7ue.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: timer,ice,replaces,outbound
User-Agent: JsSIP 3.9.0
Content-Length: 1494

v=0
o=mozilla...THIS_IS_SDPARTA-99.0 4935802565320084750 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 7E:AB:AC:A0:5B:7C:F6:6A:62:95:DC:ED:C3:D2:28:EA:44:97:26:A6:1B:13:D0:EF:66:C4:E3:CE:AB:4B:D6:36
a=group:BUNDLE 0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 63803 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 80.26.218.23
a=candidate:0 1 UDP 2122252543 192.168.0.41 63803 typ host
a=candidate:2 1 TCP 2105524479 192.168.0.41 9 typ host tcptype active
a=candidate:0 2 UDP 2122252542 192.168.0.41 63304 typ host
a=candidate:2 2 TCP 2105524478 192.168.0.41 9 typ host tcptype active
a=candidate:1 1 UDP 1686052863 80.26.218.23 63803 typ srflx raddr 192.168.0.41 rport 63803
a=candidate:1 2 UDP 1686052862 80.26.218.23 63304 typ srflx raddr 192.168.0.41 rport 63304
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:5c72374c75e7beae1681991bf9be2189
a=ice-ufrag:35dae94a
a=mid:0
a=msid:{6172abf6-4d47-411d-bc67-721b581b5a77} {0f0506fb-5c0e-41a1-ba59-96218d305cf7}
a=rtcp:63304 IN IP4 80.26.218.23
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=setup:actpass
a=ssrc:2585001976 cname:{4784dd3a-3bb1-4b62-9897-cc649f35f743}

<--- Transmitting SIP response (306 bytes) to WSS:192.168.0.1:59163 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS 9m5rnafgr7ue.invalid;rport=59163;received=192.168.0.1;branch=z9hG4bK1807946
Call-ID: 18oi83ttq3s21kl31c6a
From: <sip:webrtc_client@ha.balimelon.com>;tag=q2md5opiei
To: <sip:111@ha.balimelon.com>
CSeq: 6234 INVITE
Server: Asterisk PBX 20.8.1
Content-Length:  0


    -- Executing [111@default:1] Dial("PJSIP/webrtc_client-00000003", "PJSIP/111") in new stack
    -- Called PJSIP/111
<--- Transmitting SIP request (890 bytes) to UDP:192.168.0.222:5060 --->
INVITE sip:111@192.168.0.222:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.100:5060;rport;branch=z9hG4bKPj6364a18c-36ca-420f-8db0-67572e3a331b
From: "HASS" <sip:webrtc_client@192.168.0.100>;tag=73d0b304-6bb4-4e72-b771-2cc7e0efd319
To: <sip:111@192.168.0.222>
Contact: <sip:asterisk@192.168.0.100:5060>
Call-ID: 10a8c1e2-0435-4f6d-ae07-c29756887b45
CSeq: 24863 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, INFO
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.8.1
Content-Type: application/sdp
Content-Length:   205

v=0
o=- 220559857 220559857 IN IP4 192.168.0.100
s=Asterisk
c=IN IP4 192.168.0.100
t=0 0
m=audio 10426 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:140
a=sendrecv

<--- Received SIP response (370 bytes) from UDP:192.168.0.222:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.100:5060;rport=5060;branch=z9hG4bKPj6364a18c-36ca-420f-8db0-67572e3a331b;received=192.168.0.100
From: "HASS" <sip:webrtc_client@192.168.0.100>;tag=73d0b304-6bb4-4e72-b771-2cc7e0efd319
To: <sip:111@192.168.0.222>
Call-ID: 10a8c1e2-0435-4f6d-ae07-c29756887b45
CSeq: 24863 INVITE
Server: YATE/5.5.0
Content-Length: 0


<--- Received SIP response (469 bytes) from UDP:192.168.0.222:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.100:5060;rport=5060;branch=z9hG4bKPj6364a18c-36ca-420f-8db0-67572e3a331b;received=192.168.0.100
From: "HASS" <sip:webrtc_client@192.168.0.100>;tag=73d0b304-6bb4-4e72-b771-2cc7e0efd319
To: <sip:111@192.168.0.222>
Call-ID: 10a8c1e2-0435-4f6d-ae07-c29756887b45
CSeq: 24863 INVITE
Server: YATE/5.5.0
Contact: <sip:111@192.168.0.222:5060>
Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, OPTIONS, INFO
Content-Length: 0


<--- Transmitting SIP request (391 bytes) to UDP:192.168.0.222:5060 --->
ACK sip:111@192.168.0.222:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.100:5060;rport;branch=z9hG4bKPj6364a18c-36ca-420f-8db0-67572e3a331b
From: "HASS" <sip:webrtc_client@192.168.0.100>;tag=73d0b304-6bb4-4e72-b771-2cc7e0efd319
To: <sip:111@192.168.0.222>
Call-ID: 10a8c1e2-0435-4f6d-ae07-c29756887b45
CSeq: 24863 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 20.8.1
Content-Length:  0


  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'PJSIP/webrtc_client-00000003' status is 'CHANUNAVAIL'
<--- Transmitting SIP response (384 bytes) to WSS:192.168.0.1:59163 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/WSS 9m5rnafgr7ue.invalid;rport=59163;received=192.168.0.1;branch=z9hG4bK1807946
Call-ID: 18oi83ttq3s21kl31c6a
From: <sip:webrtc_client@ha.balimelon.com>;tag=q2md5opiei
To: <sip:111@ha.balimelon.com>;tag=d3828257-e469-4a70-8dd5-bec2ec854984
CSeq: 6234 INVITE
Server: Asterisk PBX 20.8.1
Reason: Q.850;cause=34
Content-Length:  0


<--- Received SIP request (434 bytes) from WSS:192.168.0.1:59163 --->
ACK sip:111@ha.balimelon.com SIP/2.0
Via: SIP/2.0/WSS 9m5rnafgr7ue.invalid;branch=z9hG4bK1807946
Max-Forwards: 69
To: <sip:111@ha.balimelon.com>;tag=d3828257-e469-4a70-8dd5-bec2ec854984
From: <sip:webrtc_client@ha.balimelon.com>;tag=q2md5opiei
Call-ID: 18oi83ttq3s21kl31c6a
CSeq: 6234 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: JsSIP 3.9.0
Content-Length: 0

jcolp · November 21, 2024, 12:42pm

There is nothing different between the two calls to the respective devices, aside from the dynamic aspects of SIP signaling.

sesardelaisla · November 21, 2024, 12:45pm

So this is just an unexpected behaviour by the old doorbell, right?

jcolp · November 21, 2024, 12:47pm

Based on the information given, yes.

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Response msg 404/INVITE error when calling a doorbell

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