Hi, Another issue is that your Record-Route sip-uri used "lr=on" instead of "lr". RFC 3261 defines lr-param without a value.
RFC 3261 section 16.12.1 has some loose routing and strict routing examples which may be of interest. > -----Original Message----- > From: sip-implementors-boun...@lists.cs.columbia.edu [mailto:sip- > implementors-boun...@lists.cs.columbia.edu] On Behalf Of David Cunningham > Sent: Sunday, August 23, 2015 8:27 PM > To: Joegen Baclor > Cc: sip-implementors@lists.cs.columbia.edu > Subject: Re: [Sip-implementors] R-URI on ACK > > Thank you for the explanation Joegen and apologies for my mixed terminology. > > The IP aa.aa.13.240 is our address so I wouldn't expect it to be on the route > going out. > > I think I have been misunderstanding loose routes. I had presumed they meant > the R-URI didn't change at all. > > > On 22 August 2015 at 11:25, Joegen Baclor <jbac...@ezuce.com> wrote: > > > They are correct. ACK for 200 Ok follows the rules for requests > > within a dialog. Your ACK request-uri should point to the value of > > the Contact-URI sent in the 200 Ok. > > > > >> a) Should the R-URI be changed on a reply? (as opposed to a new > > >> dialog) > > Not sure what you mean by "R-URI on a reply" . ACKs are requests. > > Responses do not have request-uris. > > > > b) Doesn't the loose routing (lr in the record-route for > > bb.bb.224.202) > >>> mean that the R-URI shouldn't be changed anyway? > >>> > >> No. The record-route or route-sets are the hops that your request > >> would > > follow. You ACK should have been contructed as follows > > > > ACK sip:+11234567...@cc.cc.13.45:5060 SIP/2.0 > > Route: <sip:aa.aa.13.240:5060;lr=on> > > Route: <sip:bb.bb.224.202:5060;lr;ftag=as1514f3e7> > > > > Take note that there were two record routes in the 200 OK. Your > > implementation must insert those two in the ACK. Your ACK appears to > > only insert the top-most route. You, therefore, have two violations here. > > Request-URI is wrong and Route-set is wrong. > > > > Joegen > > > > > > > > > > On 08/22/2015 09:01 AM, David Cunningham wrote: > > > >> Hello, > >> > >> I'm looking for some advice on a response from a SIP termination provider. > >> They send us the 200 OK below, and we send them the ACK. > >> > >> They say that our ACK is invalid because it's R-URI should contain > >> "+11234567...@cc.cc.13.45:5060" rather than > >> "+11234567...@bb.bb.224.202 :5060". > >> Presumably they say that because what they want is in the Contact > >> header in their 200 OK. > >> > >> My questions are: > >> a) Should the R-URI be changed on a reply? (as opposed to a new > >> dialog) > >> b) Doesn't the loose routing (lr in the record-route for > >> bb.bb.224.202) mean that the R-URI shouldn't be changed anyway? > >> > >> Thanks in advance for any help. > >> > >> > >> Received from provider address bb.bb.224.202: > >> > >> SIP/2.0 200 OK. > >> Via: SIP/2.0/UDP > >> aa.aa.13.240;branch=z9hG4bK02ee.83d31f591682ebd45da194d341fa23f9.0. > >> Via: SIP/2.0/UDP aa.aa.13.195:5060;rport=5060;branch=z9hG4bK4bcb23fc. > >> From: <sip:+12345678...@aa.aa.13.195>;tag=as1514f3e7. > >> To: <sip:+11234567...@ot.provider.com:5060>;tag=gK00d4fe47. > >> Call-ID: xxx. > >> CSeq: 102 INVITE. > >> Record-Route: <sip:bb.bb.224.202:5060;lr;ftag=as1514f3e7>. > >> Record-Route: <sip:aa.aa.13.240:5060;lr=on>. > >> Accept: application/sdp. > >> Contact: <sip:+11234567...@cc.cc.13.45:5060>. > >> Allow: INVITE,ACK,CANCEL,BYE,PRACK,OPTIONS. > >> Content-Length: 244. > >> Content-Disposition: session; handling=required. > >> Content-Type: application/sdp. > >> . > >> v=0. > >> o=Sonus_UAC 188128247 152711770 IN IP4 cc.cc.13.45. > >> s=SIP Media Capabilities. > >> c=IN IP4 cc.cc.13.72. > >> t=0 0. > >> m=audio 43276 RTP/AVP 0 101. > >> a=rtpmap:0 PCMU/8000. > >> a=rtpmap:101 telephone-event/8000. > >> a=fmtp:101 0-15. > >> a=sendrecv. > >> a=maxptime:20. > >> > >> > >> Sent to provider address bb.bb.224.202: > >> > >> ACK sip:+11234567...@bb.bb.224.202:5060 > >> Via: SIP/2.0/UDP > >> aa.aa.13.240;branch=z9hG4bKa205.5b9b58c0e9cb82ab7edf866d7b03a805.0 > >> Via: SIP/2.0/UDP aa.aa.13.194:5060;rport=5060;branch=z9hG4bK45d24b48 > >> Route: <sip:bb.bb.224.202:5060;lr;ftag=as2e95ab82> > >> Max-Forwards: 69 > >> From: <sip:+12345678...@aa.aa.13.194>;tag=as2e95ab82 > >> To: <sip:+11234567...@ot.provider.com:5060>;tag=gK08a9c7fd > >> Contact: <sip:+12345678...@aa.aa.13.194:5060> > >> Call-ID: xxx > >> CSeq: 102 ACK > >> User-Agent: XXX > >> Content-Length: 0 _______________________________________________ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
