Thanks !
czw., 26 sty 2023 o 16:01 Ovidiu Sas <osas(a)voipembedded.com> napisał(a):
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:+48732122479@cludotls.byoc.mypu
▒ 10:22:15.957096 │ │
│ 20│cloud.de>;tag=mDL0l1E
▒ +0.003231 │ │
│ <────────│From: "221223977" <sip:221223977@cludo.p
▒ 10:22:15.960327 │ 200 OK (SDP) │
│ │;tag=KmZjt3Fr4HgZB
▒ +0.001268 │ <────────────────────────── │
│ │Contact: <sip:+48732122479@172.23.9.70:5
▒ 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