If the REGISTER information is rewritten while receiving (outgoing) a call, it is not able to answer the call

Asterisk Asterisk SIP
1

I’ve been using Asterisk for many years. We use UAC with WebSocket signaling. When the network goes offline, it is UNREGISTERED, and when it comes back online, it is REGISTERED again.

If I disconnect from the network and reconnect while receiving (outgoing) a call, the REGSITER information will change (strictly speaking, the port number will change).
After that, even if you send an Accept signal to Asterisk, the signaling will be transported to the already UNREGISTERed port number and the signaling will not arrive properly.
Is there a setting to allow signaling to the new Port?

FYI: In AOR, max_contact = 1 and enable remove_existing.

Specific flow
A makes a call
B becomes Ringing
A disconnects from the network (UNREGISTERED port A)
A returns to the network (REGISTERED with another port B)
B accepts (sends OK+SDP signaling)
When Asterisk sends OK+SDP signal to A, Asterisk sends it to port A. (I expected to use port B)

2

You’d need to actually show configuration and the SIP signaling.

3

Thanks for your reply. I will share configuration and SIP singnaling logs

Settings is here.

ps_endpoints

                                id: aaaaaaaaaaaa
                         transport: transport-wss
                              aors: aaaaaaaaaaaa
                              auth: aaaaaaaaaaaa
                           context: sc
                          disallow: all
                             allow: opus,g722,ulaw,alaw
                      direct_media: no
             connected_line_method: NULL
               direct_media_method: NULL
     direct_media_glare_mitigation: NULL
       disable_direct_media_on_nat: NULL
                         dtmf_mode: NULL
            external_media_address: NULL
                       force_rport: yes
                       ice_support: yes
                       identify_by: NULL
                         mailboxes: NULL
                       moh_suggest: NULL
                     outbound_auth: NULL
                    outbound_proxy: NULL
                   rewrite_contact: yes
                          rtp_ipv6: NULL
                     rtp_symmetric: yes
                    send_diversion: NULL
                          send_pai: NULL
                         send_rpid: NULL
                     timers_min_se: NULL
                            timers: NULL
               timers_sess_expires: NULL
                          callerid: NULL
                  callerid_privacy: NULL
                      callerid_tag: NULL
                            100rel: NULL
                     aggregate_mwi: NULL
                  trust_id_inbound: NULL
                 trust_id_outbound: NULL
                         use_ptime: NULL
                          use_avpf: NULL
                  media_encryption: NULL
                   inband_progress: NULL
                        call_group: NULL
                      pickup_group: NULL
                  named_call_group: NULL
                named_pickup_group: NULL
              device_state_busy_at: NULL
                        fax_detect: NULL
                         t38_udptl: NULL
                      t38_udptl_ec: NULL
             t38_udptl_maxdatagram: NULL
                     t38_udptl_nat: NULL
                    t38_udptl_ipv6: NULL
                         tone_zone: NULL
                          language: NULL
               one_touch_recording: NULL
                 record_on_feature: NULL
                record_off_feature: NULL
                        rtp_engine: NULL
                    allow_transfer: NULL
                   allow_subscribe: NULL
                         sdp_owner: NULL
                       sdp_session: NULL
                         tos_audio: NULL
                         tos_video: NULL
                    sub_min_expiry: NULL
                       from_domain: NULL
                         from_user: NULL
                     mwi_from_user: NULL
                       dtls_verify: NULL
                        dtls_rekey: NULL
                    dtls_cert_file: NULL
                  dtls_private_key: NULL
                       dtls_cipher: NULL
                      dtls_ca_file: NULL
                      dtls_ca_path: NULL
                        dtls_setup: NULL
                       srtp_tag_32: NULL
                     media_address: NULL
                   redirect_method: NULL
                           set_var: NULL
                         cos_audio: NULL
                         cos_video: NULL
                   message_context: NULL
                         force_avp: NULL
      media_use_received_transport: NULL
                       accountcode: NULL
                     user_eq_phone: NULL
                   moh_passthrough: NULL
       media_encryption_optimistic: NULL
                    rpid_immediate: NULL
                 g726_non_standard: NULL
                     rtp_keepalive: NULL
                       rtp_timeout: NULL
                  rtp_timeout_hold: NULL
         bind_rtp_to_media_address: NULL
               voicemail_extension: NULL
mwi_subscribe_replaces_unsolicited: NULL
                              deny: NULL
                            permit: NULL
                               acl: NULL
                      contact_deny: NULL
                    contact_permit: NULL
                       contact_acl: NULL
                 subscribe_context: NULL
                fax_detect_timeout: NULL
                      contact_user: NULL
              preferred_codec_only: NULL
              asymmetric_rtp_codec: NULL
                          rtcp_mux: yes
                     allow_overlap: NULL
              refer_blind_progress: NULL
        notify_early_inuse_ringing: NULL
                 max_audio_streams: NULL
                 max_video_streams: NULL
                            webrtc: yes
                  dtls_fingerprint: NULL
              incoming_mwi_mailbox: NULL
                            bundle: NULL
           dtls_auto_generate_cert: NULL
           follow_early_media_fork: NULL
       accept_multiple_sdp_answers: NULL
      suppress_q850_reason_headers: NULL
              trust_connected_line: NULL
               send_connected_line: NULL
            ignore_183_without_sdp: NULL
        codec_prefs_incoming_offer: NULL
        codec_prefs_outgoing_offer: NULL
       codec_prefs_incoming_answer: NULL
       codec_prefs_outgoing_answer: NULL
                       stir_shaken: NULL
                 send_history_info: NULL
     allow_unauthenticated_options: NULL
   t38_bind_udptl_to_media_address: NULL
      geoloc_incoming_call_profile: NULL
      geoloc_outgoing_call_profile: NULL
          incoming_call_offer_pref: NULL
          outgoing_call_offer_pref: NULL
               stir_shaken_profile: NULL
              security_negotiation: NULL
               security_mechanisms: NULL
                          send_aoc: NULL
                   overlap_context: NULL

ps_aors


mysql> select * from ps_aors limit 1 \G;
*************************** 1. row ***************************
                  id: aaaaaaaaaaaa
             contact: NULL
  default_expiration: NULL
           mailboxes: NULL
        max_contacts: 1
  minimum_expiration: NULL
     remove_existing: yes
   qualify_frequency: NULL
authenticate_qualify: NULL
  maximum_expiration: NULL
      outbound_proxy: NULL
        support_path: NULL
     qualify_timeout: 9
 voicemail_extension: NULL
  remove_unavailable: NULL

pjsip.conf

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:11111
local_net=10.0.0.0/24
external_media_address=xxxxxxxxx.com
external_signaling_address= xxxxxxxxx.com

[transport-wss]
type=transport
protocol=wss
bind=0.0.0.0
external_media_address=xxxxxxxxx.com
external_signaling_address= xxxxxxxxx.com

And asterisk signaling log is below.
aaaaaaaaaaaa invite bbbbbbbbbbb. after bbbbbbbbbbb is ringing, aaaaaaaaaaaa’s network will change. Then, bbbbbbbbbbb accept the call but Asterisk send the OK(+SDP) to initial(invalid) port number.
I execute pjsip show contacts 2 times. first one is initial port, second one executed after port number is changed

ip-10-0-0-157*CLI> pjsip show contacts

  Contact:  <Aor/ContactUri..............................> <Hash....> <Status> <RTT(ms)..>
==========================================================================================

  Contact:  aaaaaaaaaaaa/sip:mt405qdc@103.5.140.163:42324;tr b73fbeb1e8 NonQual         nan
  Contact:  bbbbbbbbbbb/sip:bbbbbbbbbbb@222.222.222.222:51160; b46ce74535 NonQual         nan

Objects found: 2

<--- Received SIP request (2594 bytes) from WSS:103.5.140.163:42324 --->
INVITE sip:bbbbbbbbbbb@zzzzzzzzzzzzzzzzzzzz.com SIP/2.0
Via: SIP/2.0/WSS n0w97ugniyb7.invalid;branch=z9hG4bK7600689010000000
Max-Forwards: 69
To: <sip:bbbbbbbbbbb@zzzzzzzzzzzzzzzzzzzz.com>
From: "Flutter Client" <sip:aaaaaaaaaaaa@zzzzzzzzzzzzzzzzzzzz.com>;tag=ek4c6wutcx
Call-ID: 9e43sot1vzvuk0ln99a1
CSeq: 4353 INVITE
Contact: <sip:mt405qdc@n0w97ugniyb7.invalid;transport=WS;ob>
Content-Type: application/sdp
Session-Expires: 120
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: timer,ice,replaces,outbound
User-Agent: sc-voip-client
Content-Length: 1974

v=0
o=- 5864169624512325382 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS 922207a3-01f1-4391-8a4d-2b403c0629ed
m=audio 33330 UDP/TLS/RTP/SAVPF 111 63 9 102 0 8 13 110 126
c=IN IP4 103.5.140.163
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:1336129771 1 udp 2122260223 10.41.246.155 33330 typ host generation 0 network-id 3 network-cost 10
a=candidate:361449391 1 udp 2122129151 127.0.0.1 46778 typ host generation 0 network-id 1
a=candidate:949130100 1 udp 2122202367 ::1 39784 typ host generation 0 network-id 2
a=candidate:635115063 1 udp 1686052607 103.5.140.163 33330 typ srflx raddr 10.41.246.155 rport 33330 generation 0 network-id 3 network-cost 10
a=candidate:1799660855 1 tcp 1518149375 127.0.0.1 38619 typ host tcptype passive generation 0 network-id 1
a=candidate:1180529132 1 tcp 1518222591 ::1 53489 typ host tcptype passive generation 0 network-id 2
a=ice-ufrag:WD18
a=ice-pwd:pr8lsXI6x0jGAyv/WCfNlARI
a=ice-options:trickle renomination
a=fingerprint:sha-256 C2:FB:EA:30:FE:93:E7:40:6A:4B:8E:D0:3B:2B:71:7B:D9:56:93:44:D4:C0:3C:B1:20:08:27:32:FF:30:7F:80
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=sendrecv
a=msid:922207a3-01f1-4391-8a4d-2b403c0629ed 255be8a1-1efd-4a1f-b819-a75fc25e3d58
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:9 G722/8000
a=rtpmap:102 ILBC/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:126 telephone-event/8000
a=ssrc:1802400102 cname:Y/euPN69cenmcxYh
a=ssrc:1802400102 msid:922207a3-01f1-4391-8a4d-2b403c0629ed 255be8a1-1efd-4a1f-b819-a75fc25e3d58

<--- Transmitting SIP response (548 bytes) to WSS:103.5.140.163:42324 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS n0w97ugniyb7.invalid;rport=42324;received=103.5.140.163;branch=z9hG4bK7600689010000000
Call-ID: 9e43sot1vzvuk0ln99a1
From: "Flutter Client" <sip:aaaaaaaaaaaa@zzzzzzzzzzzzzzzzzzzz.com>;tag=ek4c6wutcx
To: <sip:bbbbbbbbbbb@zzzzzzzzzzzzzzzzzzzz.com>;tag=z9hG4bK7600689010000000
CSeq: 4353 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1721545928/cae8eca4714ca17656063fe06908a6d2",opaque="63f6b1aa5bbc9e30",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.5.0
Content-Length:  0


<--- Received SIP request (505 bytes) from WSS:103.5.140.163:42324 --->
ACK sip:bbbbbbbbbbb@zzzzzzzzzzzzzzzzzzzz.com SIP/2.0
Via: SIP/2.0/WSS n0w97ugniyb7.invalid;branch=z9hG4bK7600689010000000
Max-Forwards: 69
To: <sip:bbbbbbbbbbb@zzzzzzzzzzzzzzzzzzzz.com>;tag=z9hG4bK7600689010000000
From: "Flutter Client" <sip:aaaaaaaaaaaa@zzzzzzzzzzzzzzzzzzzz.com>;tag=ek4c6wutcx
Call-ID: 9e43sot1vzvuk0ln99a1
CSeq: 4353 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: sc-voip-client
Content-Length: 0


<--- Received SIP request (2897 bytes) from WSS:103.5.140.163:42324 --->
INVITE sip:bbbbbbbbbbb@zzzzzzzzzzzzzzzzzzzz.com SIP/2.0
Via: SIP/2.0/WSS n0w97ugniyb7.invalid;branch=z9hG4bK13185888280000000
Max-Forwards: 69
To: <sip:bbbbbbbbbbb@zzzzzzzzzzzzzzzzzzzz.com>
From: "Flutter Client" <sip:aaaaaaaaaaaa@zzzzzzzzzzzzzzzzzzzz.com>;tag=ek4c6wutcx
Call-ID: 9e43sot1vzvuk0ln99a1
CSeq: 4354 INVITE
Authorization: Digest algorithm=MD5, username="aaaaaaaaaaaa", realm="asterisk", nonce="1721545928/cae8eca4714ca17656063fe06908a6d2", uri="sip:bbbbbbbbbbb@zzzzzzzzzzzzzzzzzzzz.com", response="3312abfb1d14edabf8bce3fc82cf4ea2", opaque="63f6b1aa5bbc9e30", qop=auth, cnonce="op7s132s5r58", nc=00000001
Contact: <sip:mt405qdc@n0w97ugniyb7.invalid;transport=WS;ob>
Content-Type: application/sdp
Session-Expires: 120
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: timer,ice,replaces,outbound
User-Agent: sc-voip-client
Content-Length: 1974

v=0
o=- 5864169624512325382 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS 922207a3-01f1-4391-8a4d-2b403c0629ed
m=audio 33330 UDP/TLS/RTP/SAVPF 111 63 9 102 0 8 13 110 126
c=IN IP4 103.5.140.163
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:1336129771 1 udp 2122260223 10.41.246.155 33330 typ host generation 0 network-id 3 network-cost 10
a=candidate:361449391 1 udp 2122129151 127.0.0.1 46778 typ host generation 0 network-id 1
a=candidate:949130100 1 udp 2122202367 ::1 39784 typ host generation 0 network-id 2
a=candidate:635115063 1 udp 1686052607 103.5.140.163 33330 typ srflx raddr 10.41.246.155 rport 33330 generation 0 network-id 3 network-cost 10
a=candidate:1799660855 1 tcp 1518149375 127.0.0.1 38619 typ host tcptype passive generation 0 network-id 1
a=candidate:1180529132 1 tcp 1518222591 ::1 53489 typ host tcptype passive generation 0 network-id 2
a=ice-ufrag:WD18
a=ice-pwd:pr8lsXI6x0jGAyv/WCfNlARI
a=ice-options:trickle renomination
a=fingerprint:sha-256 C2:FB:EA:30:FE:93:E7:40:6A:4B:8E:D0:3B:2B:71:7B:D9:56:93:44:D4:C0:3C:B1:20:08:27:32:FF:30:7F:80
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=sendrecv
a=msid:922207a3-01f1-4391-8a4d-2b403c0629ed 255be8a1-1efd-4a1f-b819-a75fc25e3d58
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:9 G722/8000
a=rtpmap:102 ILBC/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:126 telephone-event/8000
a=ssrc:1802400102 cname:Y/euPN69cenmcxYh
a=ssrc:1802400102 msid:922207a3-01f1-4391-8a4d-2b403c0629ed 255be8a1-1efd-4a1f-b819-a75fc25e3d58

<--- Transmitting SIP response (369 bytes) to WSS:103.5.140.163:42324 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS n0w97ugniyb7.invalid;rport=42324;received=103.5.140.163;branch=z9hG4bK13185888280000000
Call-ID: 9e43sot1vzvuk0ln99a1
From: "Flutter Client" <sip:aaaaaaaaaaaa@zzzzzzzzzzzzzzzzzzzz.com>;tag=ek4c6wutcx
To: <sip:bbbbbbbbbbb@zzzzzzzzzzzzzzzzzzzz.com>
CSeq: 4354 INVITE
Server: Asterisk PBX 20.5.0
Content-Length:  0


    -- Executing [bbbbbbbbbbb@sc:1] Set("PJSIP/aaaaaaaaaaaa-00000000", "_somevar=") in new stack
    -- Executing [bbbbbbbbbbb@sc:2] Set("PJSIP/aaaaaaaaaaaa-00000000", "PJSIP_HEADER(add,X-myheader=") in new stack
    -- Executing [bbbbbbbbbbb@sc:3] Dial("PJSIP/aaaaaaaaaaaa-00000000", "PJSIP/bbbbbbbbbbb/sip:bbbbbbbbbbb@222.222.222.222:51160;rinstance=972466e6e9cb53ec,60,Tt") in new stack
    -- Called PJSIP/bbbbbbbbbbb/sip:bbbbbbbbbbb@222.222.222.222:51160;rinstance=972466e6e9cb53ec
<--- Transmitting SIP request (1096 bytes) to UDP:222.222.222.222:51160 --->
INVITE sip:bbbbbbbbbbb@222.222.222.222:51160;rinstance=972466e6e9cb53ec SIP/2.0
Via: SIP/2.0/UDP 111.111.111.111(asterisk server id):19240;rport;branch=z9hG4bKPj6784bb03-12a1-410a-90be-c84ec9b439b9
From: "Flutter Client" <sip:aaaaaaaaaaaa@10.0.0.157>;tag=68dba13d-7742-49bb-883e-ecc286d2668c
To: <sip:bbbbbbbbbbb@222.222.222.222;rinstance=972466e6e9cb53ec>
Contact: <sip:asterisk@111.111.111.111(asterisk server id):19240>
Call-ID: b9a49a86-05f7-4a0d-a099-78dcd58674a9
CSeq: 12526 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Type: application/sdp
Content-Length:   342

v=0
o=- 2001074942 2001074942 IN IP4 111.111.111.111(asterisk server id)
s=Asterisk
c=IN IP4 111.111.111.111(asterisk server id)
t=0 0
m=audio 18198 RTP/AVP 107 9 0 8 101
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

<--- Received SIP response (368 bytes) from UDP:222.222.222.222:51160 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 111.111.111.111(asterisk server id):19240;rport=19240;branch=z9hG4bKPj6784bb03-12a1-410a-90be-c84ec9b439b9
To: <sip:bbbbbbbbbbb@222.222.222.222;rinstance=972466e6e9cb53ec>
From: "Flutter Client" <sip:aaaaaaaaaaaa@10.0.0.157>;tag=68dba13d-7742-49bb-883e-ecc286d2668c
Call-ID: b9a49a86-05f7-4a0d-a099-78dcd58674a9
CSeq: 12526 INVITE
Content-Length: 0


<--- Received SIP response (715 bytes) from UDP:222.222.222.222:51160 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 111.111.111.111(asterisk server id):19240;rport=19240;branch=z9hG4bKPj6784bb03-12a1-410a-90be-c84ec9b439b9
Contact: <sip:bbbbbbbbbbb@222.222.222.222:51160;transport=UDP>
To: <sip:bbbbbbbbbbb@222.222.222.222;rinstance=972466e6e9cb53ec>;tag=ff0f7d3d
From: "Flutter Client" <sip:aaaaaaaaaaaa@10.0.0.157>;tag=68dba13d-7742-49bb-883e-ecc286d2668c
Call-ID: b9a49a86-05f7-4a0d-a099-78dcd58674a9
CSeq: 12526 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.6.1 v2.10.19.9
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 0


    -- PJSIP/bbbbbbbbbbb-00000001 is ringing
<--- Transmitting SIP response (568 bytes) to WSS:103.5.140.163:42324 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WSS n0w97ugniyb7.invalid;rport=42324;received=103.5.140.163;branch=z9hG4bK13185888280000000
Call-ID: 9e43sot1vzvuk0ln99a1
From: "Flutter Client" <sip:aaaaaaaaaaaa@zzzzzzzzzzzzzzzzzzzz.com>;tag=ek4c6wutcx
To: <sip:bbbbbbbbbbb@zzzzzzzzzzzzzzzzzzzz.com>;tag=f993168a-f79d-4775-91d4-28dfa2761f56
CSeq: 4354 INVITE
Server: Asterisk PBX 20.5.0
Contact: <sip:10.0.0.157:19240;transport=ws>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length:  0


  == WebSocket connection from '103.5.140.163:50524' for protocol 'sip' accepted using version '13'
<--- Received SIP request (645 bytes) from WSS:103.5.140.163:50524 --->
REGISTER sip:zzzzzzzzzzzzzzzzzzzz.com SIP/2.0
Via: SIP/2.0/WSS n0w97ugniyb7.invalid;branch=z9hG4bK577585579
Max-Forwards: 69
To: <sip:aaaaaaaaaaaa@zzzzzzzzzzzzzzzzzzzz.com>
From: "Flutter Client" <sip:aaaaaaaaaaaa@zzzzzzzzzzzzzzzzzzzz.com>;tag=rc2iv7n5lj
Call-ID: 6s7rg510zjp3x7fag58l7q
CSeq: 3 REGISTER
Contact: <sip:mt405qdc@n0w97ugniyb7.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:dfb036da-f5ec-4f23-a3fb-6204860e871b>";expires=600
Expires: 600
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: path,gruu,outbound
User-Agent: sc-voip-client
Content-Length: 0


<--- Transmitting SIP response (535 bytes) to WSS:103.5.140.163:50524 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS n0w97ugniyb7.invalid;rport=50524;received=103.5.140.163;branch=z9hG4bK577585579
Call-ID: 6s7rg510zjp3x7fag58l7q
From: "Flutter Client" <sip:aaaaaaaaaaaa@zzzzzzzzzzzzzzzzzzzz.com>;tag=rc2iv7n5lj
To: <sip:aaaaaaaaaaaa@zzzzzzzzzzzzzzzzzzzz.com>;tag=z9hG4bK577585579
CSeq: 3 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1721545942/d999d1ae363aaf19931057a7e60b23a8",opaque="2bb5e96c560fbdcd",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.5.0
Content-Length:  0


<--- Received SIP request (937 bytes) from WSS:103.5.140.163:50524 --->
REGISTER sip:zzzzzzzzzzzzzzzzzzzz.com SIP/2.0
Via: SIP/2.0/WSS n0w97ugniyb7.invalid;branch=z9hG4bK1432920139
Max-Forwards: 69
To: <sip:aaaaaaaaaaaa@zzzzzzzzzzzzzzzzzzzz.com>
From: "Flutter Client" <sip:aaaaaaaaaaaa@zzzzzzzzzzzzzzzzzzzz.com>;tag=rc2iv7n5lj
Call-ID: 6s7rg510zjp3x7fag58l7q
CSeq: 4 REGISTER
Authorization: Digest algorithm=MD5, username="aaaaaaaaaaaa", realm="asterisk", nonce="1721545942/d999d1ae363aaf19931057a7e60b23a8", uri="sip:zzzzzzzzzzzzzzzzzzzz.com", response="e49ccd88de5a9812c13a62424c41bfd4", opaque="2bb5e96c560fbdcd", qop=auth, cnonce="waufzfszqjnh", nc=00000001
Contact: <sip:mt405qdc@n0w97ugniyb7.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:dfb036da-f5ec-4f23-a3fb-6204860e871b>";expires=600
Expires: 600
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: path,gruu,outbound
User-Agent: sc-voip-client
Content-Length: 0


    -- Added contact 'sip:mt405qdc@103.5.140.163:50524;transport=ws;x-ast-orig-host=n0w97ugniyb7.invalid:0' to AOR 'aaaaaaaaaaaa' with expiration of 600 seconds
    -- Removed contact 'sip:mt405qdc@103.5.140.163:42324;transport=ws;x-ast-orig-host=n0w97ugniyb7.invalid:0' from AOR 'aaaaaaaaaaaa' due to remove existing
  == Contact aaaaaaaaaaaa/sip:mt405qdc@103.5.140.163:42324;transport=ws;x-ast-orig-host=n0w97ugniyb7.invalid:0 has been deleted
<--- Transmitting SIP response (503 bytes) to WSS:103.5.140.163:50524 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS n0w97ugniyb7.invalid;rport=50524;received=103.5.140.163;branch=z9hG4bK1432920139
Call-ID: 6s7rg510zjp3x7fag58l7q
From: "Flutter Client" <sip:aaaaaaaaaaaa@zzzzzzzzzzzzzzzzzzzz.com>;tag=rc2iv7n5lj
To: <sip:aaaaaaaaaaaa@zzzzzzzzzzzzzzzzzzzz.com>;tag=z9hG4bK1432920139
CSeq: 4 REGISTER
Date: Sun, 21 Jul 2024 07:12:22 GMT
Contact: <sip:mt405qdc@n0w97ugniyb7.invalid;transport=ws>;expires=599
Expires: 600
Server: Asterisk PBX 20.5.0
Content-Length:  0


ip-10-0-0-157*CLI> pjsip show contacts

  Contact:  <Aor/ContactUri..............................> <Hash....> <Status> <RTT(ms)..>
==========================================================================================

  Contact:  aaaaaaaaaaaa/sip:mt405qdc@103.5.140.163:50524;tr a426cc2bae NonQual         nan
  Contact:  bbbbbbbbbbb/sip:bbbbbbbbbbb@222.222.222.222:51160; b46ce74535 NonQual         nan

Objects found: 2

<--- Received SIP response (1186 bytes) from UDP:222.222.222.222:51160 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.111.111(asterisk server id):19240;rport=19240;branch=z9hG4bKPj6784bb03-12a1-410a-90be-c84ec9b439b9
Require: timer
Contact: <sip:bbbbbbbbbbb@222.222.222.222:51160;transport=UDP>
To: <sip:bbbbbbbbbbb@222.222.222.222;rinstance=972466e6e9cb53ec>;tag=ff0f7d3d
From: "Flutter Client" <sip:aaaaaaaaaaaa@10.0.0.157>;tag=68dba13d-7742-49bb-883e-ecc286d2668c
Call-ID: b9a49a86-05f7-4a0d-a099-78dcd58674a9
CSeq: 12526 INVITE
Session-Expires: 1800;refresher=uac
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.6.1 v2.10.19.9
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 390

v=0
o=Z 0 254719044 IN IP4 222.222.222.222
s=Z
c=IN IP4 222.222.222.222
t=0 0
m=audio 51858 RTP/AVP 107 9 0 8 18 3 101 96
a=rtpmap:107 opus/48000/2
a=fmtp:107 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/48000
a=fmtp:101 0-16
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=sendrecv
a=rtcp-mux

       > 0x7fa87005d7c0 -- Strict RTP learning after remote address set to: 222.222.222.222:51858
       > 0x7fa87005d7c0 -- Strict RTP switching to RTP target address 222.222.222.222:51858 as source
<--- Transmitting SIP request (452 bytes) to UDP:222.222.222.222:51160 --->
ACK sip:bbbbbbbbbbb@222.222.222.222:51160 SIP/2.0
Via: SIP/2.0/UDP 111.111.111.111(asterisk server id):19240;rport;branch=z9hG4bKPj0de1b201-3a2b-4043-b447-8cbd9b0f026e
From: "Flutter Client" <sip:aaaaaaaaaaaa@10.0.0.157>;tag=68dba13d-7742-49bb-883e-ecc286d2668c
To: <sip:bbbbbbbbbbb@222.222.222.222;rinstance=972466e6e9cb53ec>;tag=ff0f7d3d
Call-ID: b9a49a86-05f7-4a0d-a099-78dcd58674a9
CSeq: 12526 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length:  0


    -- PJSIP/bbbbbbbbbbb-00000001 answered PJSIP/aaaaaaaaaaaa-00000000
       > 0x7fa8700fa5f0 -- Strict RTP learning after remote address set to: 103.5.140.163:33330
<--- Transmitting SIP response (1673 bytes) to WSS:103.5.140.163:42324 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS n0w97ugniyb7.invalid;rport=42324;received=103.5.140.163;branch=z9hG4bK13185888280000000
Call-ID: 9e43sot1vzvuk0ln99a1
From: "Flutter Client" <sip:aaaaaaaaaaaa@zzzzzzzzzzzzzzzzzzzz.com>;tag=ek4c6wutcx
To: <sip:bbbbbbbbbbb@zzzzzzzzzzzzzzzzzzzz.com>;tag=f993168a-f79d-4775-91d4-28dfa2761f56
CSeq: 4354 INVITE
Server: Asterisk PBX 20.5.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:10.0.0.157:19240;transport=ws>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 120;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   976

v=0
o=- 733046534 4 IN IP4 10.0.0.157
s=Asterisk
c=IN IP4 10.0.0.157
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0
m=audio 19648 UDP/TLS/RTP/SAVPF 111 9 0 8 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 CD:3F:13:CA:68:7D:06:E7:C5:3F:BB:0A:60:F8:01:56:A2:F2:2D:B5:5A:C0:22:E7:EA:54:03:98:A9:9A:16:11
a=ice-ufrag:649af0ec5bb2e1fa7ef1fa0e15addea3
a=ice-pwd:65bb283f543360153107091f175fa6f2
a=candidate:Ha00009d 1 UDP 2130706431 10.0.0.157 19648 typ host
a=candidate:S234af81e 1 UDP 1694498815 111.111.111.111(asterisk server id) 19648 typ srflx raddr 10.0.0.157 rport 19648
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:1941942047 cname:751ed823-d3e8-435a-968d-d8203c3b0002
a=msid:f06678d1-d13f-42cb-8867-cfddb4ca5a88 613fd87f-194d-4b5c-9575-6369705efca4
a=rtcp-fb:* transport-cc
a=mid:0

    -- Channel PJSIP/bbbbbbbbbbb-00000001 joined 'simple_bridge' basic-bridge <58506356-8ea0-43e7-9719-ed74f2dc23e1>
    -- Channel PJSIP/aaaaaaaaaaaa-00000000 joined 'simple_bridge' basic-bridge <58506356-8ea0-43e7-9719-ed74f2dc23e1>
[2024-07-21 16:12:27] ERROR[246907]: res_http_websocket.c:567 ws_safe_read: Error reading from web socket: Connection reset by peer
<--- Transmitting SIP response (1673 bytes) to WSS:103.5.140.163:42324 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS n0w97ugniyb7.invalid;rport=42324;received=103.5.140.163;branch=z9hG4bK13185888280000000
Call-ID: 9e43sot1vzvuk0ln99a1
From: "Flutter Client" <sip:aaaaaaaaaaaa@zzzzzzzzzzzzzzzzzzzz.com>;tag=ek4c6wutcx
To: <sip:bbbbbbbbbbb@zzzzzzzzzzzzzzzzzzzz.com>;tag=f993168a-f79d-4775-91d4-28dfa2761f56
CSeq: 4354 INVITE
Server: Asterisk PBX 20.5.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:10.0.0.157:19240;transport=ws>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 120;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   976

v=0
o=- 733046534 4 IN IP4 10.0.0.157
s=Asterisk
c=IN IP4 10.0.0.157
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0
m=audio 19648 UDP/TLS/RTP/SAVPF 111 9 0 8 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 CD:3F:13:CA:68:7D:06:E7:C5:3F:BB:0A:60:F8:01:56:A2:F2:2D:B5:5A:C0:22:E7:EA:54:03:98:A9:9A:16:11
a=ice-ufrag:649af0ec5bb2e1fa7ef1fa0e15addea3
a=ice-pwd:65bb283f543360153107091f175fa6f2
a=candidate:Ha00009d 1 UDP 2130706431 10.0.0.157 19648 typ host
a=candidate:S234af81e 1 UDP 1694498815 111.111.111.111(asterisk server id) 19648 typ srflx raddr 10.0.0.157 rport 19648
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:1941942047 cname:751ed823-d3e8-435a-968d-d8203c3b0002
a=msid:f06678d1-d13f-42cb-8867-cfddb4ca5a88 613fd87f-194d-4b5c-9575-6369705efca4
a=rtcp-fb:* transport-cc
a=mid:0

[2024-07-21 16:12:27] WARNING[246864]: pjproject: <?>: 	     tsx0x7fa870102658 .Error sending Response msg 200/INVITE/cseq=4354 (tdta0x7fa87c00dc38): Unknown Error (PJ_EUNKNOWN)
[2024-07-21 16:12:27] ERROR[246864]: iostream.c:552 ast_iostream_close: SSL_shutdown() failed: error:00000005:lib(0)::reason(5), Underlying BIO error: Bad file descriptor
  == WebSocket connection from '103.5.140.163:42324' forcefully closed due to fatal write error
    -- Channel PJSIP/aaaaaaaaaaaa-00000000 left 'simple_bridge' basic-bridge <58506356-8ea0-43e7-9719-ed74f2dc23e1>
  == Spawn extension (sc, bbbbbbbbbbb, 3) exited non-zero on 'PJSIP/aaaaaaaaaaaa-00000000'
    -- Channel PJSIP/bbbbbbbbbbb-00000001 left 'simple_bridge' basic-bridge <58506356-8ea0-43e7-9719-ed74f2dc23e1>
<--- Transmitting SIP request (476 bytes) to UDP:222.222.222.222:51160 --->
BYE sip:bbbbbbbbbbb@222.222.222.222:51160 SIP/2.0
Via: SIP/2.0/UDP 111.111.111.111(asterisk server id):19240;rport;branch=z9hG4bKPj17824cb7-996f-49a5-9d89-84949a7ab440
From: "Flutter Client" <sip:aaaaaaaaaaaa@10.0.0.157>;tag=68dba13d-7742-49bb-883e-ecc286d2668c
To: <sip:bbbbbbbbbbb@222.222.222.222;rinstance=972466e6e9cb53ec>;tag=ff0f7d3d
Call-ID: b9a49a86-05f7-4a0d-a099-78dcd58674a9
CSeq: 12527 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length:  0


<--- Received SIP response (467 bytes) from UDP:222.222.222.222:51160 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 111.111.111.111(asterisk server id):19240;rport=19240;branch=z9hG4bKPj17824cb7-996f-49a5-9d89-84949a7ab440
Contact: <sip:bbbbbbbbbbb@222.222.222.222:51160;transport=UDP>
To: <sip:bbbbbbbbbbb@222.222.222.222;rinstance=972466e6e9cb53ec>;tag=ff0f7d3d
From: "Flutter Client" <sip:aaaaaaaaaaaa@10.0.0.157>;tag=68dba13d-7742-49bb-883e-ecc286d2668c
Call-ID: b9a49a86-05f7-4a0d-a099-78dcd58674a9
CSeq: 12527 BYE
User-Agent: Z 5.6.1 v2.10.19.9
Content-Length: 0


ip-10-0-0-157*CLI>
4

Wait, you’re expecting the new SIP registration to override the active in-call for the 200 OK from Asterisk to the client? That’s not how SIP works. The registration is only used once to get the URI to start the call. Subsequently that URI is used, unless the in-dialog Contact is updated.

5

I got it. Thanks for your answer!

6

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