I've opened the captured traffic with Wireshark and I've found out
that Kamailio is actually forwarding the call with a double SDP
header. ngrep was showing only a partial header. I think this happens
because when forwarding the call I use rtpengine_offer() and not
rtpengine_manage(). I've tried using rtpengine_manage() and I get only
one SDP header but with the (wrong) private IP address. I've tried
also rtpengine_delete() before rtpengine_offer() but there's always a
double header. Ideas? Still digging..
Thanks!
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): Sonus_UAC 157372 901736 IN IP4 44.55.73.157
Session Name (s): SIP Media Capabilities
Connection Information (c): IN IP4 172.30.0.156
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 30154 RTP/AVP 0 18 101
Media Attribute (a): maxptime:20
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:18 G729/8000
Media Attribute (a): fmtp:18 annexb=no
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-15
Media Attribute (a): sendrecv
Media Attribute (a): rtcp:30155
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): Sonus_UAC 157372 901736 IN IP4 44.55.73.157
Session Name (s): SIP Media Capabilities
Connection Information (c): IN IP4 22.33.177.156
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 30358 RTP/AVP 0 18 101
Media Attribute (a): maxptime:20
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:18 G729/8000
Media Attribute (a): fmtp:18 annexb=no
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-15
Media Attribute (a): sendrecv
Media Attribute (a): rtcp:30359
[Generated Call-ID: 241993330_65984054@44.55.73.157]
On Wed, Oct 11, 2023 at 12:29 PM Leonardo Arena <rna...@gmail.com> wrote:
>
> Hi all,
>
> I have the following setup:
>
> Phone <-> private net using TCP transport <-> (172.30.0.156) Kamailio
> v5.6.3 + RTPEngine v9.3.1 (22.33.177.156) <-> public net using UDP
> transport <-> Asterisk (22.33.178.77)
>
> Phone is registered on Kamailio and forward calls on no answer or when
> ext is unregistered, to Asterisk's custom IVR (nodejs app + CouchDB),
> using failure_route() [1].
>
> The problem is that when the none answer the phone and the call is
> redirected to Asterisk, Kamailio announce its private IP address in
> the SDP connection endpoint, hence there is no audio [2]. From
> RTPEngine debug logs, it seems that RTPEngine replies with the correct
> SDP connection endpoint IP [3]. See also RTPEngine interfaces
> definition [4].
>
> What am I missing? Why Kamailio is offering its private IP address in
> the SDP connection, instead of its public address when forwarding the
> call? If comment "route(SDP_MANAGE_IN)" in route[IN], Kamailio uses
> the correct IP endpoint address in the SDP offer when forwarding the
> call to Asterisk. But of course this breaks audio when someone answer
> the phone. It seems that Kamailio is storing the initial SDP offer
> settings and does not update it upon RTPEngine's request, even though
> I have added "replace-session-connection" in route[SDP_OFFER_IVR].
>
> Thanks for any help!
>
> leo
>
>
> [1]
> route[IN] {
> if (!lookup('location')) {
> t_on_reply("SDP_MANAGE");
> route(TOIVR);
> } else {
> xlog("L_INFO", "Route(IN) Req $mi $rm From <$fu> To <$tu> RURI
> <$ru>\n");
> record_route();
> t_on_reply("SDP_MANAGE");
> route(SDP_MANAGE_IN);
> t_on_failure("TOIVR");
> t_relay();
> exit;
> }
> }
>
> route[SDP_MANAGE] {
> if (has_body("application/sdp")) {
> rtpengine_manage();
> }
> }
>
> route[SDP_MANAGE_IN] {
> if (has_body("application/sdp")) {
> rtpengine_manage("direction=public direction=private via-branch=1");
> }
> }
>
>
> ...
>
>
> route[SDP_OFFER_IVR] {
> if (has_body("application/sdp")) {
> rtpengine_offer("direction=public direction=public
> replace-session-connection via-branch=1");
> }
> }
>
> failure_route[TOIVR] {
> if (t_is_canceled()) {
> exit;
> }
> append_branch();
> route(SDP_OFFER_IVR);
> # using t_relay_to so that UDP is used with the correct IP address
> if (!t_relay_to("xh.voip.net")) {
> xlog("L_INFO", "No answer - Failure Route(TOIVR) Req $mi $rm
> From <$fu> To <$tu> RURI <$ru> failed\n");
> t_reply("500", "Unable to route");
> } else {
> xlog("L_INFO", "No answer - Failure Route(TOIVR) Req $mi $rm
> From <$fu> To <$tu> RURI <$ru>\n");
> }
> }
>
>
> [2]
> U 2023/10/11 09:05:43.797393 22.33.177.156:5060 -> 22.33.176.8:5060 #3
> SIP/2.0 180 Ringing.
> Via: SIP/2.0/UDP
> 22.33.176.8;branch=z9hG4bK2cb2.e559c4a326678f69ae9b4656be7bc36c.0.
> Via: SIP/2.0/UDP
> 22.33.176.20;branch=z9hG4bK2cb2.2920561c845ee5279051359f40cb7c0f.0.
> Via: SIP/2.0/UDP 44.55.73.157:5060;branch=z9hG4bK0cBdccef20c0d8bfefb.
> Record-Route: <sip:172.30.0.156;transport=TCP;r2=on;lr=on;ftag=gK0c1bb12e>.
> Record-Route: <sip:22.33.177.156;r2=on;lr=on;ftag=gK0c1bb12e>.
> Record-Route: <sip:22.33.176.8;lr=on;ftag=gK0c1bb12e;did=7ffe.26c8>.
> Record-Route: <sip:22.33.176.20;lr=on;ftag=gK0c1bb12e>.
> From: "Unavailable" <sip:013335553222@44.55.73.157>;tag=gK0c1bb12e.
> To: <sip:8555000033@22.33.176.20>;tag=660096088.
> Call-ID: 474758100_116775912@44.55.73.157.
> CSeq: 704680 INVITE.
> Contact: <sip:+18555000033@172.30.96.2:11815;transport=TCP>.
> Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY,
> REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE.
> User-Agent: Yealink SIP-T31P 124.86.0.40.
> Allow-Events: talk,hold,conference,refer,check-sync.
> Content-Length: 0.
> .
>
>
> U 2023/10/11 09:06:03.751585 22.33.177.156:5060 -> 22.33.178.77:5060 #4
> INVITE sip:+18555000033@172.30.96.2:11815;transport=TCP SIP/2.0.
> Record-Route: <sip:22.33.177.156;lr=on;ftag=gK0c1bb12e>.
> Record-Route: <sip:22.33.176.8;lr=on;ftag=gK0c1bb12e;did=7ffe.26c8>.
> Record-Route: <sip:22.33.176.20;lr=on;ftag=gK0c1bb12e>.
> Via: SIP/2.0/UDP
> 22.33.177.156;branch=z9hG4bK2cb2.afb825e30e294872c72c5e21ef26f3f3.1.
> Via: SIP/2.0/UDP
> 22.33.176.8;branch=z9hG4bK2cb2.e559c4a326678f69ae9b4656be7bc36c.0.
> Via: SIP/2.0/UDP
> 22.33.176.20;branch=z9hG4bK2cb2.2920561c845ee5279051359f40cb7c0f.0.
> Via: SIP/2.0/UDP 44.55.73.157:5060;branch=z9hG4bK0cBdccef20c0d8bfefb.
> f: "Unavailable" <sip:013335553222@44.55.73.157>;tag=gK0c1bb12e.
> t: <sip:8555000033@22.33.176.20>.
> i: 474758100_116775912@44.55.73.157.
> CSeq: 704680 INVITE.
> Max-Forwards: 32.
> Allow:
> INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH.
> Accept: application/sdp, application/isup, application/dtmf,
> application/dtmf-relay, multipart/mixed.
> m: "Unavailable" <sip:013335553222@44.55.73.157:5060>.
> Remote-Party-ID: "Unavailable"
> <sip:013335553222@44.55.73.157:5060>;privacy=off.
> k: timer,100rel,precondition.
> Session-Expires: 1800.
> Min-SE: 90.
> l: 604.
> Content-Disposition: session; handling=required.
> c: application/sdp.
> .
> v=0.
> o=Sonus_UAC 525653 764118 IN IP4 44.55.73.157.
> s=SIP Media Capabilities.
> c=IN IP4 172.30.0.156.
> t=0 0.
> m=audio 30030 RTP/AVP 0 18 101.
> a=maxptime:20.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=n
>
>
>
>
> [3]
> Oct 11 09:06:03 voipsipr7 daemon.info rtpengine[35061]: INFO:
> [474758100_116775912@44.55.73.157]: [control] Received command 'offer'
> from 127.0.0.1:39942
> Oct 11 09:06:03 voipsipr7 daemon.debug rtpengine[35061]: DEBUG:
> [474758100_116775912@44.55.73.157]: [control] Dump for 'offer' from
> 127.0.0.1:39942: { "supports": [ "load limit" ], "sdp": "v=0^M
> o=Sonus_UAC 5
> 25653 764118 IN IP4 44.55.73.157^M s=SIP Media Capabilities^M c=IN IP4
> 44.55.78.60^M t=0 0^M m=audio 61610 RTP/AVP 0 18 101^M a=rtpmap:0
> PCMU/8000^M a=rtpmap:18 G729/8000^M a=fmtp:18 annexb=no^M a=rtpmap:101
> telephone-event/8000^M a=fmtp:101 0-15^M a=sendrecv^M a=maxptime:20^M
> ", "direction": [ "public", "public" ], "replace": [
> "session-connection" ], "call-id": "474758100_116775912@44.55.73.157
> ...
> Oct 11 09:06:03 voipsipr7 daemon.debug rtpengine[35061]: DEBUG:
> [474758100_116775912@44.55.73.157]: ... ", "via-branch":
> "z9hG4bK2cb2.e559c4a326678f69ae9b4656be7bc36c.0", "received-from": [
> "IP4", "22.33.176
> .8" ], "from-tag": "gK0c1bb12e", "command": "offer" }
>
>
> ...
>
>
> Oct 11 09:06:03 voipsipr7 daemon.info rtpengine[35061]: INFO:
> [474758100_116775912@44.55.73.157]: [control] Replying to 'offer' from
> 127.0.0.1:39942 (elapsed time 0.000638 sec)
> Oct 11 09:06:03 voipsipr7 daemon.debug rtpengine[35061]: DEBUG:
> [474758100_116775912@44.55.73.157]: [control] Response dump for
> 'offer' to 127.0.0.1:39942: { "sdp": "v=0^M o=Sonus_UAC 525653 764118
> IN IP4 207
> .223.73.157^M s=SIP Media Capabilities^M c=IN IP4 22.33.177.156^M t=0
> 0^M m=audio 30068 RTP/AVP 0 18 101^M a=maxptime:20^M a=rtpmap:0
> PCMU/8000^M a=rtpmap:18 G729/8000^M a=fmtp:18 annexb=no^M a=rtpmap:101
> teleph
> one-event/8000^M a=fmtp:101 0-15^M a=sendrecv^M a=rtcp:30069^M ",
> "result": "ok" }
> Oct 11 09:06:03 voipsipr7 daemon.info /usr/sbin/kamailio[35261]: INFO:
> <script>: No answer - Failure Route(TOIVR) Req 4 INVITE From
> <sip:013335553222@44.55.73.157> To <sip:8555000033@22.33.176.20> RURI
> <sip:
> +18555000033@172.30.96.2:11815;transport=TCP>
> Oct 11 09:06:03 voipsipr7 daemon.info rtpengine[35061]: INFO:
> [474758100_116775912@44.55.73.157]: [control] Received command
> 'answer' from 127.0.0.1:53245
> Oct 11 09:06:03 voipsipr7 daemon.debug rtpengine[35061]: DEBUG:
> [474758100_116775912@44.55.73.157]: [control] Dump for 'answer' from
> 127.0.0.1:53245: { "supports": [ "load limit" ], "sdp": "v=0^M o=-
> 525653 7
> 64120 IN IP4 22.33.178.77^M s=Asterisk^M c=IN IP4 22.33.178.77^M t=0
> 0^M m=audio 11264 RTP/AVP 0 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:101
> telephone-event/8000^M a=fmtp:101 0-16^M a=ptime:20^M a=maxptime:150^M
> a=sendrecv^M m=audio 0 RTP/AVP 0 18 101^M ", "call-id":
> "474758100_116775912@44.55.73.157", "received-from": [ "IP4",
> "22.33.178.77" ], "from-tag": "gK0c1bb12e", "to-tag":
> "a21c53b6-ca33-43b ...
> Oct 11 09:06:03 voipsipr7 daemon.debug rtpengine[35061]: DEBUG:
> [474758100_116775912@44.55.73.157]: ... 4-a585-af9b568a5ecb",
> "command": "answer" }
>
>
> ...
>
>
> Oct 11 09:06:03 voipsipr7 daemon.info rtpengine[35061]: INFO:
> [474758100_116775912@44.55.73.157]: [control] Replying to 'answer'
> from 127.0.0.1:53245 (elapsed time 0.000915 sec)
> Oct 11 09:06:03 voipsipr7 daemon.debug rtpengine[35061]: DEBUG:
> [474758100_116775912@44.55.73.157]: [control] Response dump for
> 'answer' to 127.0.0.1:53245: { "sdp": "v=0^M o=- 525653 764120 IN IP4
> 22.33.178
> .77^M s=Asterisk^M c=IN IP4 22.33.177.156^M t=0 0^M m=audio 30056
> RTP/AVP 0 101^M a=maxptime:150^M a=rtpmap:0 PCMU/8000^M a=rtpmap:101
> telephone-event/8000^M a=fmtp:101 0-16^M a=sendrecv^M a=rtcp:30057^M
> a=ptime
> :20^M m=audio 0 RTP/AVP 0 18 101^M ", "result": "ok" }
> Oct 11 09:06:03 voipsipr7 daemon.debug rtpengine[35061]: DEBUG:
> [474758100_116775912@44.55.73.157]: [core] Forward to sink endpoint:
> 22.33.178.77:11264 (RTP seq 0 TS 0)
>
>
> [4]
> Interface definition in RTPEngine:
> interface =
> public/22.33.177.156;public/22:33:201:105::156;private/172.30.0.156
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