[OpenSIPS-Users] Drouting relay issue

Volkan Oransoy voransoy at gmail.com
Wed May 11 10:52:04 UTC 2022


I think the screenshot has been discarded by the mailman. The sip traffic is as follows. The proxy tries to retransmit and fails after three more attempts.

──────────┬───────── ──────────┬───────── ──────────┬─────────
 10:47:32.603828 │ INVITE (SDP) │ │
 +0.000347 │ ──────────────────────────> │ │
 10:47:32.604175 │ 100 Giving it a try │ │
 +0.001474 │ <────────────────────────── │ │
 10:47:32.605649 │ │ INVITE (SDP) │
 +0.490742 │ │ ──────────────────────────> │
 10:47:33.096391 │ │ INVITE (SDP) │
 +1.001859 │ │ ────────────────────────>>> │
 10:47:34.098250 │ │ INVITE (SDP) │
 +1.953642 │ │ ────────────────────────>>> │
 10:47:36.051892 │ │ INVITE (SDP) │
 +1.603582 │ │ ────────────────────────>>> │
 10:47:37.655474 │ 408 Request Timeout │ │
 +0.001615 │ <────────────────────────── │ │
 10:47:37.657089 │ ACK │ │
 │ ──────────────────────────> │ │

Volkan Oransoy
On 11 May 2022 11:14 +0100, Volkan Oransoy <voransoy at gmail.com>, wrote:
> Hi all,
>
> I have an interesting issue with one of my test setups. I have a simple routing script which gets the gateway id directly from the header originating from a Freeswith box. The system finds and sets the request URL as anticipated. But even if I can see the request on the proxy, I can't see the traffic on the destination. Interestingly, the same proxy can register to the same destination with uac_registrant as a UAC. And I can receive calls from the same destination. Is there anything missing to route this traffic correctly?
>
> Thanks in advance.
>
> route[to_gateway] {
>         if ( route_to_gw($hdr(X-GWID)) ) {
>                 route(relay);
>         }
> }
> route[relay] {
>         if (is_method("INVITE")) {
>                 t_on_branch("per_branch_ops");
>                 t_on_reply("handle_nat");
>                 t_on_failure("failure");
>         }
>         if (!t_relay()) {
>                 send_reply(500,"Internal Error");
>         }
>         exit;
> }
>
> The database structure is as follows;
>
> opensips=# select * from dr_gateways;
>  id | gwid | type |           address            | strip | pri_prefix | attrs | probe_mode | state | socket | description
> ----+------+------+------------------------------+-------+------------+-------+------------+-------+--------+-------------
>   6 | 5    |    2 | sip:testgw.bulutfon.net:5060 |     0 |            |       |          0 |     0 |        | 5
>
> Here is the INVITE request sent to the destination, which fails as in the screenshot.
>
> INVITE sip:905551234567 at testgw.bulutfon.net:5060 SIP/2.0
> Record-Route: <sip:2.2.2.2;lr;ftag=1jcyr93emrjDQ>
> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK372.d935ecb2.0
> Via: SIP/2.0/UDP 1.1.1.1:6080;received=1.1.1.1;rport=6080;branch=z9hG4bK99mtDaevmN7Nj
> Max-Forwards: 67
> From: "+908508850000" <sip:908508850000 at testgw.bulutfon.net:5060>;tag=1jcyr93emrjDQ
> To: <sip:905551234567 at testgw.bulutfon.net:5060>
> Call-ID: aefb8c60-426b-123b-8ca8-82a722ba4f75
> CSeq: 51067506 INVITE
> Contact: <sip:mod_sofia at 1.1.1.1:6080>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
> Supported: timer, path, replaces
> Allow-Events: talk, hold, conference, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 621
> Remote-Party-ID: "+908508850000" <sip:+908508850000 at testgw.bulutfon.net:5060>;party=calling;screen=yes;privacy=off
>
> v=0
> o=FreeSWITCH 1651210438 1651210439 IN IP4 1.1.1.1
> s=FreeSWITCH
> c=IN IP4 1.1.1.1
> t=0 0 ...
>
> <Screen Shot 2022-04-29 at 15.13.21.png>
> --
>
> Volkan Oransoy
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