AFAIK the only way to "match" reg and invite (by sip design) is to use separate auth data for each reg. Client could use random port for UAC's and e.g. 5060 for UAS, so you can't really rely on contact vs src host:port matching.
2014-11-28 22:07 GMT+03:00 Luis Azedo luis@2600hz.com:
Hi,
when using sip.js websocket client i got this sequence
REGISTER sip:mysip.com SIP/2.0 Via: SIP/2.0/WSS 1pds95dm7slp.invalid;branch=z9hG4bK1393343 Max-Forwards: 70 To: sip:1096@mysip.com From: sip:1096@mysip.com;tag=99i9ka9951 Call-ID: 1agj9ujebn5k2006qvosg7 CSeq: 82 REGISTER Authorization: Digest algorithm=MD5, username="1096", realm="mysip.com", nonce="VHiXFFR4legH+zdoSro9zybQQclvFtcR", uri="sip:mysip.com", response="2d9843d09d94f295d0ebfa323233783c" Contact: sip:n7o5lk6u@1pds95dm7slp.invalid ;transport=ws;reg-id=1;+sip.instance="urn:uuid:d68a53c4-37e7-4f5c-87f2-68d7cca14b93";expires=600 Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE Supported: path,gruu,100rel,outbound User-Agent: SIP.js/0.6.3 Content-Length: 0
SIP/2.0 200 OK Via: SIP/2.0/WSS sg6h3at19h29.invalid;branch=z9hG4bK4876180;rport=65377;received=X.X.X.X To: sip:1096@mysip.com;tag=78e678a5264e254e660865f4df24edf3.e69b From: sip:1096@mysip.com;tag=j39lvle827 Call-ID: c0ofajiui4u2vtem662rdh CSeq: 86 REGISTER Contact: sip:8umlvk34@sg6h3at19h29.invalid ;transport=ws;expires=600;pub-gruu="sip:1096@mysip.com ;gr=urn:uuid:876de649-281f-4f7f-9a3e-7b2a7e11f3cc";temp-gruu=" sip:uloc-5478bf26-4504-1-7fd4326@mysip.com ;gr";+sip.instance="urn:uuid:876de649-281f-4f7f-9a3e-7b2a7e11f3cc";reg-id=1 Content-Length: 0
INVITE sip:2134@mysip.com SIP/2.0 Via: SIP/2.0/WSS sg6h3at19h29.invalid;branch=z9hG4bK1752419 Max-Forwards: 70 To: sip:2134@mysip.com From: sip:1096@mysip.com;tag=6vii55tc02 Call-ID: i3hndju3tgsge7ctnvo2 CSeq: 4782 INVITE Contact: sip:1096@mysip.com ;gr=urn:uuid:876de649-281f-4f7f-9a3e-7b2a7e11f3cc Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE Content-Type: application/sdp Supported: gruu,100rel,outbound User-Agent: SIP.js/0.6.3 Content-Length: 2585
i issued kamctl show to view the registered user
AOR:: 1096@mysip.com Contact:: sip:8umlvk34@sg6h3at19h29.invalid;transport=wsQ= Expires:: 177 Callid:: c0ofajiui4u2vtem662rdh Cseq:: 86 User-agent:: SIP.js/0.6.3 Received:: sip:X.X.X.X:65377;transport=WS State:: CS_SYNC Flags:: 0 Cflag:: 6 Socket:: tls:Y.Y.Y.Y:8443 Methods:: 927 Ruid:: uloc-5478bf26-4504-1 Instance:: urn:uuid:876de649-281f-4f7f-9a3e-7b2a7e11f3cc Reg-Id:: 1 Last-Keepalive:: 1417200609 Last-Modified:: 1417200609
what is the best way to check if the INVITE is from a registered user and how to get usrloc properties for the registered user ?
i tried this
reg_fetch_contacts("location", "$fu", "caller"); $var(i) = 0; $var(found) = 0; $var(contact) = $(ct{nameaddr.uri}); while($var(found) == 0 && $var(i) < $(ulc(caller=>count))) { xlog("L_INFO", "$ci|log| checking against $(ulc(caller=>addr)[$var(i)])"); if($var(contact) == $(ulc(caller=>addr)[$var(i)])) { $var(found) = 1; $var(caller_idx) = $var(i); $var(userlocid) = $(ulc(caller=>ruid)[$var(i)]); } else $var(i) = $var(i) + 1; }
the problem i see is that the INVITE Contact doesn't match the one registered.
is this a problem with sip.js or am i missing something in kamailio.cfg or should kamailio save the contact info as sip.js is sending?
Thank you
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