Just wanted to let everybody know that flowroute has fixed the delay issue by reconfiguring their gateways. The route field no longer causes delays and I experienced excellent voice quality. Their tech support was extremely helpful and I can now highly recommend flowroute in conjunction with sipX.
Marcello -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Marcello Manzardo Sent: Thursday, March 04, 2010 1:53 PM To: 'Scott Lawrence'; 'M. Ranganathan' Cc: [email protected] Subject: Re: [sipx-users] Delayed call setup with Flow Route - Wrong field in INVITE Thank you very much for your replies. I forwarded this info about RFC 3261 and Route header to flowroute. Has anybody successfully used flowroute as an ITSP? -----Original Message----- From: Scott Lawrence [mailto:[email protected]] Sent: Thursday, March 04, 2010 1:24 PM To: M. Ranganathan Cc: [email protected]; [email protected] Subject: Re: [sipx-users] Delayed call setup with Flow Route - Wrong field in INVITE On Thu, 2010-03-04 at 16:09 -0500, M. Ranganathan wrote: > On Thu, Mar 4, 2010 at 3:57 PM, Marcello Manzardo <[email protected]> > wrote: > > I am using 4.0.4-017289 ecs-centos5 > > > > > > > > I have setup flowroute ITSP but experiencing about a 1 min delay until a > > call goes through. > > > > Tech support from flowroute pointed to a wrong field in the SIP INVITE that > > is generated by sipX. > > > > The field in question is the “Route” field that is not supposed to be there > > according to RFC 3261 (section 16.12.1). It is that route field that is > > causing the delay according to tech support. > > > That is not correct. For SipXbridge to work, it is a requirement that > RFC 3261 be supported. It is quite legal in Rfc 3261 to put a loose > route header on an INVITE. > > > > I am quite sure there is something else wrong. > > INVITE sip:[email protected];user=phone SIP/2.0 > > Route: <sip:70.167.153.130:5060;transport=udp;lr> More specifically - that Route header points to exactly the same proxy that resolving sip.flowroute.com finds, so there's not a thing wrong with it. The section that flowroute cited is just a list of examples; section 16.12 says : 2. The proxy will inspect the URI in the topmost Route header field value. If it indicates this proxy, the proxy removes it from the Route header field (this route node has been reached). that's exactly what that Route header does, so they should remove it and resolve the request URI to route the call. _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/ _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
