What is see is that ACK is being not recognized in any living session
Feb 1 14:46:37 localhost kamailio[1463]: DEBUG: {1 udp 63075809 ACK
b39129b0-1cd9-123c-718c-001851b691ff} <core> [core/receive.c:3 79]:
receive_msg(): preparing to run routing scripts...
Feb 1 14:46:37 localhost kamailio[1463]: DEBUG: {1 udp 63075809 ACK
b39129b0-1cd9-123c-718c-001851b691ff} sl [sl_funcs.c:443]:
sl_filter_ACK(): too late to be a local ACK!
Feb 1 14:46:37 localhost kamailio[1463]: DEBUG: {1 udp 63075809 ACK
b39129b0-1cd9-123c-718c-001851b691ff} tm [t_lookup.c:497]:
t_lookup_request(): start searching: hash=383, isACK=1
Feb 1 14:46:37 localhost kamailio[1463]: DEBUG: {1 udp 63075809 ACK
b39129b0-1cd9-123c-718c-001851b691ff} tm [t_lookup.c:455]:
matching_3261(): RFC3261 transaction matching failed - via branch
[z9hG4bKyUr334QyvjKNB]
Feb 1 14:46:37 localhost kamailio[1463]: DEBUG: {1 udp 63075809 ACK
b39129b0-1cd9-123c-718c-001851b691ff} tm [t_lookup.c:675]:
t_lookup_request(): no transaction found
Feb 1 14:46:37 localhost kamailio[1463]: DEBUG: {1 udp 63075809 ACK
b39129b0-1cd9-123c-718c-001851b691ff} <core>
[core/parser/parse_addr_spec.c:185]: parse_to_param(): add param:
tag=QcU134KD12NSc
Feb 1 14:46:37 localhost kamailio[1463]: DEBUG: {1 udp 63075809 ACK
b39129b0-1cd9-123c-718c-001851b691ff} <core>
[core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header
reached, state=29
Feb 1 14:46:37 localhost kamailio[1463]: DEBUG: {1 udp 63075809 ACK
b39129b0-1cd9-123c-718c-001851b691ff} siptrace [siptrace_hep.c:496]:
pipport2su(): the port string is 5060
śr., 1 lut 2023 o 11:01 Krzysztof Drewicz <[email protected]>
napisał(a):
>
> Thanks !
>
>
> czw., 26 sty 2023 o 16:01 Ovidiu Sas <[email protected]> napisał(a):
> >
> > Here's a quick guide on how to capture and decrypt encrypted traffic:
> > https://voipembedded.wordpress.com/2021/03/22/troubleshooting-kamailio-encrypted-sip-traffic/
>
> Now i see my mistake -
>
> │SIP/2.0 200 OK
> 172.23.210.210:5060 172.23.9.70:5060
> 1.2.3.22:5061 │Via: SIP/2.0/UDP 172.23.210.210;received1
> 8.197.58.44:57301───┬───────── ──────────┬─────────
> ──────────┬───────── │72.23.210.210;rport=5060;branch=z9hG4bKm─
> ▒ 10:22:15.656278 │ INVITE (SDP) │
> │ │a4jBUK71BH
> ▒ +0.000730 │ ──────────────────────────> │
> │ │Record-Route: <sip:10.72.42.1:5060;r2=on
> ▒ 10:22:15.657008 │ 100 trying -- your call is │
> │ │tag=KmZjt3Fr4HgZB;lr>,<sip:18.197.58.44:
> ▒ +0.004722 │ <────────────────────────── │
> │ │61;r2=on;transport=tls;ftag=KmZjt3Fr4HgZ
> ▒ 10:22:15.661730 │ │
> │ IN│lr>,<sip:1.2.3.22:5061;transport=tl
> ▒ +0.027368 │ │
> │ ─────────│r2=on;lr=on;ftag=KmZjt3Fr4HgZB>,<sip:172
> ▒ 10:22:15.689098 │ │
> │ 1│3.9.70;r2=on;lr=on;ftag=KmZjt3Fr4HgZB>
> ▒ +0.267998 │ │
> │ <────────│To: <sip:[email protected]
> ▒ 10:22:15.957096 │ │
> │ 20│cloud.de>;tag=mDL0l1E
> ▒ +0.003231 │ │
> │ <────────│From: "221223977" <sip:[email protected]
> ▒ 10:22:15.960327 │ 200 OK (SDP) │
> │ │;tag=KmZjt3Fr4HgZB
> ▒ +0.001268 │ <────────────────────────── │
> │ │Contact: <sip:[email protected]:5
> ▒ 10:22:15.961595 │ ACK │
> │ │0;alias=18.197.58.44~5061~3>
> ▒ +0.000164 │ ──────────────────────────> │
> │ │Call-ID: c565792b-1cb4-123c-708c-001851b
> ▒ 10:22:15.961759 │ ACK │
> │ │1ff
> ▒ +0.020839 │ ──────────────────────────> │
> │ │CSeq: 63067878 INVITE
> ▒ 10:22:15.982598 │ BYE │
> │ │Allow: INVITE, ACK, CANCEL, BYE, OPTIONS
> │ +0.003088 │ ──────────────────────────> │
> │ │INFO
> │ 10:22:15.985686 │ │
> │ │Supported: norefersub, timer
> │ +0.471166 │ │
> │ ─────────│Accept: application/sdp
> │ 10:22:16.456852 │ │
> │ 20│x-inin-cnv: b8c9493d-0b01-4f57-b0b6-673d
> │ +0.001765 │ │
> │ <<<──────│788f5a
> │ 10:22:16.458617 │ ACK │
> │ │Session-Expires: 3600;refresher=uac
> │ +0.000088 │ ──────────────────────────> │
> │ │Require: timer
> │ 10:22:16.458705 │ ACK │
> │
>
>
>
> ACKs from - plain RTP don't travel from kamailio and rtpengien to sRTP
> part there are two ACKs and they don't go to TLS+sRTP party.
>
> what i do:
>
> (..)
>
> # Wrapper for relaying requests
> route[RELAY] {
> handle_ruri_alias();
> record_route();
>
> if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {
> if (!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH");
> xlog("L_ERR","ACK I:$var(i) branch_route \n");
> }
> (..)
>
> and
>
>
> # Handle requests within SIP dialogs
> route[WITHINDLG] {
> if (!has_totag()) return;
>
> if ($si=="PBXIP" && $(ru{param.value,alias})!=$null) {
> xlog("L_INFO","[R-WINDLG_INPBX]: jestem w srodku
> $si:$sp $ru\n");
> route(RTPMANAGE);
> route(RELAY);
> exit;
> }
>
> if (loose_route()) {
> if ( is_method("NOTIFY") ) {
> record_route();
> }
>
> route(RTPMANAGE);
> route(RELAY);
> exit;
> }
>
> if ( is_method("ACK|BYE") ) {
> route(RTPMANAGE);
> route(RELAY);
> }
>
> simply copy and paste from many examples lying in the internet - but
> well - i'm stuck here.
>
> The sRTP and TLS part - work as dedigned :)
> scenario when call comes from TLS/sRTP party - towards the unencrytped
> part - also works.
>
> BR
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