[SR-Users] UPDATE relayed from wrong socket

Sergiu Pojoga pojogas at gmail.com
Fri Oct 12 16:53:09 CEST 2018


Hi Daniel,

Certainly, below find the initial INVITE and the subsequent UPDATE, as
received by Kamailio at 65.xx.xx.167. If those aren't sufficient, let me know
and if it's ok with you, I'll send the full pcap in private.

The dilemma in my mind is whether the UPDATE should have a pre-set Route
header, similar to how the INVITE has.

2018/10/11 12:34:57.339306 65.xx.xx.172:5060 -> 65.xx.xx.167:5060



INVITE sip:238 at 65.xx.xx.161:64877;rinstance=8a315091627cc10b SIP/2.0



Via: SIP/2.0/UDP 65.xx.xx.172:5060;branch=z9hG4bK694382a1



Max-Forwards: 70



Route:
<sip:65.xx.xx.167;lr;received=sip:65.xx.xx.161:64877;r2=on>,<sip:xx.xx.xx.167:5070;lr;received=sip:65.xx.xx.161:64877;r2=on>


From: "Robert" <sip:226 at mypbx.net>;tag=as0ecef1c4


To: <sip:238 at 65.xx.xx.161:64877;rinstance=8a315091627cc10b>



Contact: <sip:226 at 65.xx.xx.172:5060>



Call-ID: 1e82197b42f0173b25e70759753d4210 at mypbx.net


CSeq: 102 INVITE




                Supported: replaces, timer, path





Content-Type: application/sdp



Content-Length: 386


2018/10/11 12:35:06.096457 65.xx.xx.172:5060 -> 65.xx.xx.167:5060



UPDATE sip:238 at 10.17.0.35:64877;alias=65.xx.xx.161~64877~1 SIP/2.0



Via: SIP/2.0/UDP 65.xx.xx.172:5060;branch=z9hG4bK34fab05c



Max-Forwards: 70



From: "Robert" <sip:226 at mypbx.net>;tag=as0ecef1c4


To: <sip:238 at 65.xx.xx.161:64877;rinstance=8a315091627cc10b>;tag=6467b07f



Contact: <sip:226 at 65.xx.xx.172:5060>



Call-ID: 1e82197b42f0173b25e70759753d4210 at mypbx.net


CSeq: 103 UPDATE



Content-Length: 0

Much obliged.

On Fri, Oct 12, 2018 at 9:38 AM Daniel-Constantin Mierla <miconda at gmail.com>
wrote:

> Hello,
>
> you hve to provide the sip traffic for this case, the screenshot doesn't
> show the sip headers used for routing in this case, therefore grab the sip
> traffic for all sip messages in such scenarion, either ngrep output or pcap
> file, and send it over to see if some headers are missing or not set
> properly.
> Cheers,
> Daniel
>
> On 11.10.18 21:03, Sergiu Pojoga wrote:
>
> Hi ppl,
>
> I have this problem with call transfer, may be someone can help.
>
> The phone to the far right is registered with the Registrar to the far
> left using two PATH headers (trespassing two proxy ports, 5070 then 5060).
>
> As you can see in the graph below, after receiving the UPDATE request,
> Kamailio relays it further from port 5060, I expect it to be from 5070 just
> like the dialog forming INVITE and the CANCEL afterwards.
>
> [image: image.png]
>
> The UPDATE has a to-tag, but unlike the original INVITE - it has no Route
> header!???
>
> route[*WITHINDLG*] {
> if (!has_totag()) return;
> if (loose_route()) {
> route(DLGURI);
> if (is_method("BYE")) {
>
> ...
>
> }
> else if ( is_method("ACK") ) {
> route(NATMANAGE);
> }
> else if ( is_method("NOTIFY") ) {
> record_route();
> }
>
> route(RELAY);
> exit;
> }
>
> if ( is_method("ACK") ) {
>
> ...
>
> }
>
> # handle UPDATE method for in-dialog requests
> if (is_method("*UPDATE*")) {
> route(DLGURI);
> record_route();
> route(RELAY);
> }
> }
>
> Thanks in advance.
>
>
>
> _______________________________________________
> Kamailio (SER) - Users Mailing Listsr-users at lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
> --
> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda
> Kamailio World Conference -- www.kamailioworld.com
> Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com
>
>
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