I’m coming back to this very old question as we have still not resolved this issue with Juniper.
Are you aware of any RFC section that mandates, that the VIA headers IP+port should match the effective (transport) IP+port. Or how a UAS should interpret a mismatch? I’m seeing the following behavior of SER/OpenSER. REGISTER request is received from UAC via TCP. VIA header contains an IP + port. TCP source port of REGISTER request does NOT match port in VIA header: -> UAS ignores CONTACT header and uses effective (TCP source) port + IP of REGISTER request as contact address for future INVITE messages it sends to the UAC I understand WHY this is done (i.e. to make UAC behind NAT work). However I wonder if this specific behavior is based on a particular RFC recommendation. The reason for my question is, that the above scenario happens with the SIP-Alg of our Juniper firewall. However the firewall rejects the INVITEs from the UAS. Juniper acknowledged, that the port in the rewritten VIA header of the REGISTER request does match the effective TCP port used to send it to the UAS, but they do not consider this being in contradiction to any RFC. Best regards, Joachim >> On 13/02/15 17:55, Joachim Büchse wrote: >>> Good day, >>> >>> I’m experiencing some problems with our VoiP providers handling of REGISTER >>> requests. We are using a Gigaset PRO N720 as UAC behind a Juniper SSG 140 >>> with SIP-Alg enabled. This setup kind of works with UDP but our provider >>> wants us to use TCP. With TCP enforced incoming calls don’t work. I’ve done >>> some wire tracing and to me it seems that the providers configuration is to >>> blame, but then - there are many RFCs out there and many NAT and UAC bug >>> workarounds. Anyway, I wanted to get the opinion of “the" experts about how >>> the requests send to the UAS SHOULD be correctly interpreted. >>> >>> >>> The REGISTER requests/responses look like this (outside of the firewall): >>> >>> Protocol TCP! >>> client port 19091 <-> server port 5060 >>> >>> REGISTER sip:pbx.peoplefone.ch <sip:pbx.peoplefone.ch> SIP/2.0 >>> Via: SIP/2.0/TCP >>> 212.126.160.92:6717;rport;branch=z9hG4bKc1375589832468de63a719eac31156ec >>> From: "Michel" <sip:90780408...@pbx.peoplefone.ch >>> <sip:90780408...@pbx.peoplefone.ch>>;tag=2153084485 >>> To: "Michel" <sip:90780408...@pbx.peoplefone.ch >>> <sip:90780408...@pbx.peoplefone.ch>> >>> Call-ID: 2825358480@10_10_128_10 >>> CSeq: 1 REGISTER >>> Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp >>> <sip:90780408050@212.126.160.92:6717;transport=tcp>> >>> Max-Forwards: 70 >>> User-Agent: N720-DM-PRO/70.089.00.000.000 >>> Expires: 180 >>> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY >>> Content-Length: 0 >>> >>> SIP/2.0 401 Unauthorized >>> Via: SIP/2.0/TCP >>> 212.126.160.92:6717;rport=19091;branch=z9hG4bKc1375589832468de63a719eac31156ec >>> From: "Michel" <sip:90780408...@pbx.peoplefone.ch >>> <sip:90780408...@pbx.peoplefone.ch>>;tag=2153084485 >>> To: "Michel" <sip:90780408...@pbx.peoplefone.ch >>> <sip:90780408...@pbx.peoplefone.ch>>;tag=a0440f545f39b2694d387b475a5f6bc9.b8fc >>> Call-ID: 2825358480@10_10_128_10 >>> CSeq: 1 REGISTER >>> WWW-Authenticate: Digest realm="pbx.peoplefone.ch >>> <http://pbx.peoplefone.ch/>", nonce="VNqJBVTah9m57ZGGs8b5XCTM3GyaExDy" >>> Server: kamailio (3.2.1 (x86_64/linux)) >>> Content-Length: 0 >>> >>> REGISTER sip:pbx.peoplefone.ch <sip:pbx.peoplefone.ch> SIP/2.0 >>> Via: SIP/2.0/TCP >>> 212.126.160.92:6717;rport;branch=z9hG4bK9c27afea96e2af4baab2f8d144a588e0 >>> From: "Michel" <sip:90780408...@pbx.peoplefone.ch >>> <sip:90780408...@pbx.peoplefone.ch>>;tag=2153084485 >>> To: "Michel" <sip:90780408...@pbx.peoplefone.ch >>> <sip:90780408...@pbx.peoplefone.ch>> >>> Call-ID: 2825358480@10_10_128_10 >>> CSeq: 2 REGISTER >>> Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp >>> <sip:90780408050@212.126.160.92:6717;transport=tcp>> >>> Authorization: Digest username="90780408050", realm="pbx.peoplefone.ch >>> <http://pbx.peoplefone.ch/>", uri="sip:pbx.peoplefone.ch >>> <sip:pbx.peoplefone.ch>", nonce="VNqJBVTah9m57ZGGs8b5XCTM3GyaExDy", >>> response="764f371a08d258157a249f8d1b852514" >>> Max-Forwards: 70 >>> User-Agent: N720-DM-PRO/70.089.00.000.000 >>> Expires: 180 >>> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY >>> Content-Length: 0 >>> >>> SIP/2.0 200 OK >>> Via: SIP/2.0/TCP >>> 212.126.160.92:6717;rport=19091;branch=z9hG4bK9c27afea96e2af4baab2f8d144a588e0 >>> From: "Michel" <sip:90780408...@pbx.peoplefone.ch >>> <sip:90780408...@pbx.peoplefone.ch>>;tag=2153084485 >>> To: "Michel" <sip:90780408...@pbx.peoplefone.ch >>> <sip:90780408...@pbx.peoplefone.ch>>;tag=a0440f545f39b2694d387b475a5f6bc9.6bda >>> Call-ID: 2825358480@10_10_128_10 >>> CSeq: 2 REGISTER >>> Contact: <sip:90780408050@212.126.160.92:6717;transport=tcp >>> <sip:90780408050@212.126.160.92:6717;transport=tcp>>;q=0;expires=180;received="sip:212.126.160.92:19091;transport=TCP >>> <sip:212.126.160.92:19091;transport=TCP>" >>> Server: kamailio (3.2.1 (x86_64/linux)) >>> Content-Length: 0 >>> >>> >>> The ip:port the firewall is sending those requests from is ip >>> 212.126.160.92 port 19091. So this does NOT match the port from the Contact >>> header. For TCP this seems rather logical to me, as one cant be listening >>> on a TCP port and use it for sending at the same time. The UAC closes this >>> “register connection” with TCP FIN after the register, and so does the >>> firewall. >>> >>> However unfortunately subsequent requests from the provider (ie UAS) come >>> in on port 19091 (not port 6717 from the Contact header) and the firewall >>> simply drops them. >>> >>> Observations: >>> - the server does NOT include received=212.126.160.92 in the Via of the >>> reponse. According to RFC3581 this is mandatory when rport is present in >>> the request, so this is probably an error in the server. >>> - the server does include >>> received="sip:212.126.160.92:19091;transport=TCP >>> <sip:212.126.160.92:19091;transport=TCP>” in the Contact of the response. I >>> didnt see this in any RFC (which means nothing;-) but it could be an error. >>> - after the client received the 200 OK it closes the TCP connection. >>> - the server tries several times to re-contact the client (incoming TCP >>> SYN). However not on port 6717 (defined in the Contact header) but on port >>> 19091 (where the REGISTER came from). >>> >>> RFC3581 defines special behaviour when “rport” is defined in the request >>> (i.e. response should go to the same port the request came from) - however >>> it’s not so clear if this should apply to subsequent (INVITE/OPTIONS) >>> requests from the server to the client. Those are strictly spoken not >>> replies (or are they?). >>> >>> RFC5626 defines that a “proxy” should keep track of the flows over which it >>> received a registration and send requests over the same flow. It is not >>> clear if RFC5626 should be applied. The RFC5626 defines that a UAC includes >>> an “ob” parameter in the Contact field if it whishes further requests over >>> the same flow. Also the RFC mandates a client to add a "reg-id=x" in the >>> Contact field. Both are not the case here, so in short I think RFC5626 >>> should NOT be applied. In which case conecting to the originating port >>> (instead of the Contact port) would be a server error. >>> >>> So in short and if I interpret the RFCs correctly, the client is reachable >>> and should be contacted on >>> Transport: TCP >>> IP: 212.126.160.92 >>> Port: 6717 >>> >>> >>> If anyone who lives and breathes SIP could enlighten me if the UAS is right >>> to call back on 19091 instead of 6717 I would really appreciate it;-)) >>> >>> Best regards, >>> Joachim >>> >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierla >> http://twitter.com/#!/miconda <http://twitter.com/#!/miconda> - >> http://www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda> >> Kamailio World Conference, May 27-29, 2015 >> Berlin, Germany - http://www.kamailioworld.com >> <http://www.kamailioworld.com/> >> >
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users