[Serusers] Please help : how to disable losse routing ???

['Jan Janak']

On 16-09 13:02, Martin Koenig wrote:
> Hello,
> 
> Ngrep:
> 
> Hello,
> 
> ngrep dump:
> 
> #
> U 2004/09/16 12:42:01.791105 4.5.6.156:5060 -> 1.2.3.67:5070
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 1.2.3.67:5070;branch=z9hG4bK0a65d941.
> From: "tpl-dev" <sip:t02 at example.com>;tag=0006283e0a68003a5a99de34-50ace3d6.
> To: <sip:00358400589247 at example.com>;tag=1c16103.
> Call-ID: 0006283e-0a680037-0dc88799-351df346 at 1.2.3.67.
> CSeq: 102 INVITE.
> Contact: <sip:00358400589247 at 11.22.33.66;user=phone>.
> Record-Route:
> <sip:+358400589247 at 1.2.3.68:5070;ftag=0006283e0a68003a5a99de34-50ace3d6;lr>,
> <sip:+358400589247 at 1.2.3.69:5070;ftag=0006283e0a68003a5a99de34-50ace3d6;lr>,
> <sip:00358400589247 at 4.5.6.156;ftag=0006283e0a68003a5a99de34-50ace3d6;lr>.
> Supported: em,timer,replaces,100rel.
> Allow:
> REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE.
> Server: Audiocodes-Sip-Gateway-tpl_voip_gw1/v.4.20.354.574.
> Content-Type: application/sdp.
> Content-Length: 208.
> .
> v=0.
> o=AudiocodesGW 49132 44576 IN IP4 11.22.33.66.
> s=Phone-Call.
> c=IN IP4 11.22.33.66.
> t=0 0.
> m=audio 6000 RTP/AVP 8 96.
> a=rtpmap:8 pcma/8000.
> a=rtpmap:96 telephone-event/8000.
> a=fmtp:96 0-15.
> a=ptime:20.
> 
> #
> U 2004/09/16 12:42:02.008739 1.2.3.67:5070 -> 4.5.6.156:5060
> ACK
> sip:00358400589247 at 4.5.6.156:5060;ftag=0006283e0a68003a5a99de34-50ace3d6;lr
> SIP/2.0.
> Via: SIP/2.0/UDP 1.2.3.67:5070;branch=z9hG4bK2c7403dc.
> From: "tpl-dev" <sip:t02 at example.com>;tag=0006283e0a68003a5a99de34-50ace3d6.
> To: <sip:00358400589247 at example.com>;tag=1c16103.
> Call-ID: 0006283e-0a680037-0dc88799-351df346 at 1.2.3.67.
> CSeq: 102 ACK.
> User-Agent: CSCO/7.
> Route:
> <sip:+358400589247 at 1.2.3.69:5070;ftag=0006283e0a68003a5a99de34-50ace3d6;lr>,
> <sip:+358400589247 at 1.2.3.68:5070;ftag=0006283e0a68003a5a99de34-50ace3d6;lr>,
> <sip:00358400589247 at 11.22.33.66:5060;user=phone>.
> Content-Length: 0.
> .
> 
> #
> U 2004/09/16 12:43:55.175413 1.2.3.67:5070 -> 4.5.6.156:5060
> BYE
> sip:00358400589247 at 4.5.6.156:5060;ftag=0006283e0a68003a5a99de34-50ace3d6;lr
> SIP/2.0.
> Via: SIP/2.0/UDP 1.2.3.67:5070;branch=z9hG4bK6c7c16c5.
> From: "tpl-dev" <sip:t02 at example.com>;tag=0006283e0a68003a5a99de34-50ace3d6.
> To: <sip:00358400589247 at example.com>;tag=1c16103.
> Call-ID: 0006283e-0a680037-0dc88799-351df346 at 1.2.3.67.
> CSeq: 103 BYE.
> User-Agent: CSCO/7.
> Content-Length: 0.
> Route:
> <sip:+358400589247 at 1.2.3.69:5070;ftag=0006283e0a68003a5a99de34-50ace3d6;lr>,
> <sip:+358400589247 at 1.2.3.68:5070;ftag=0006283e0a68003a5a99de34-50ace3d6;lr>,
> <sip:00358400589247 at 11.22.33.66:5060;user=phone>.
> .
> end dump
> 
> 1.2.3.67 is the cisco 7960
> 4.5.6.156 is the local proxy
> 1.2.3.68 and .69 are proxies along the way
> 11.22.33.66 is the pstn gateway
> 
> As you can see, the cisco receives an ok with record route header field, the
> last entry 4.5.6.156:5060. Contact 11.22.33.66:5060 is the Gateway.
> 
> In my understanding for proper loose routing the ACK and also the following
> bye should be sent like this:
> 
> ACK sip:00358400589247 at 11.22.33.66:5060 (the final destination)
> Route:
> <sip:00358400589247 at 4.5.6.156:5060;lr>,<sip:+358400589247 at 1.2.3.69:5070;lr>,
> <sip:+358400589247 at 1.2.3.68:5070;lr>
> 
> Which is the reverse order of the record route from the 200 ok.
> 
> But instead the Cisco sends:
> 
> ACK sip:00358400589247 at 4.5.6.156:5060 (the next hop)
> Route:
> <sip:+358400589247 at 1.2.3.69:5070;lr>,<sip:+358400589247 at 1.2.3.68:5070;lr>,
> <sip:00358400589247 at 11.22.33.66:5060> (the final destination).
> 
> This looks like strict routing to me.

  Yes that is correct, it implements strict routing (properly), there is
  nothing wrong with it, the proxy server can detect it and handle the
  message properly. Loose routers are backwards compatible with strict
  routers.

    Jan.