[Serusers] Revisted Error: force_rtp_proxy2: can't extract bodyfrom the message

[Java Rockx]

--- "Greger V. Teigre" <greger at teigre.com> wrote:

> Hi,
> I've been following this thread as I have experienced the same problems 
> myself.  When I get incoming calls (both from Cisco IP-PSTN gateway and from 
> other SIP phones) to a Grandstream behind symmetric NAT, the messages you 
> have noted can be seen in the log when hanging up.
> 
> I was not certain as to the conclusion you ended with.  Do you use the 
> filter:
>                 if (!(search("^Content-Length:\ 0")) {
>                         force_rtp_proxy();
>                 };
> 
> to avoid the errors?  I have been thinking about testing on method and not 
> call force on BYE and ACK.  Have you tried this?
> 
> I also saw your question on RFC compliance and the Sonus equipment:  In 
> order to make Grandstream phones register properly when using STUN behind 
> symmetric NAT, I had to patch nathelper with the rport != port of received 
> address check.  (I use 0.8.14 and I guess you already have the patch with 
> the development version).  The reason is that Grandstream attempts to 
> rewrite the address using STUN even though it correctly detects a symmetric 
> NAT.   I have seen that this was introduced in a new firmware not long ago 
> (release notes).  This pussles me as sources I have seen claims this to be 
> invalid behavior (which seems correct to me).
> 
> Best regards,
> Greger
> 
> 
> Java Rockx wrote:
> > Hi All.
> >
> > I've hacked my ser.cfg but can someone comment on why I would be
> > recieving a "200 OK" with a
> >
> > The change I made to my onreply_route is below. The only thing I can
> > see about these messages versus others is that the CSeq says "CSeq:
> > {some digits} BYE" with "Content-Length: 0".
> >
> > So for these messages I'm just not calling force_rtp_proxy().
> >
> > I don't know if this is a symptom of my Grandstream BT100 only of if
> > other ATAs or IP phones do this.
> >
> > Regards,
> > Paul
> >
> > onreply_route[1] {
> >
> >
> >        if (isflagset(2) && status =~ "(183)|2[0-9][0-9]") {
> >
> >
> >                fix_nated_contact();
> >
> >
> >                if (!(search("^Content-Length:\ 0")) {
> >                        force_rtp_proxy();
> >                };
> >
> >
> >        } else if (nat_uac_test("1")) {
> >
> >
> >                fix_nated_contact();
> >        };
> > }
> >
> >
> > --- Java Rockx <javarockx at yahoo.com> wrote:
> >
> >> Hi all.
> >>
> >> I've got nathelper and rtpproxy working very well with my firewall.
> >> However I do still recieve these messages in my syslog. I am only
> >> catching 183 and 2xx errors in my onreply_route so I'm very
> >> confused how to prevent these errors.
> >>
> >> I'm using ser-0.8.99-dev12. Can anyone give me some advise?
> >> Cheers,
> >> Paul
> >>
> >> NOTE: The SIP message that caused these errors is at the bottom of
> >> this message.
> >>
> >>  0(27011) ERROR: extract_body: message body has length zero
> >>  0(27011) ERROR: force_rtp_proxy2: can't extract body from the
> >>  message 0(27011) ERROR: on_reply processing failed
> >>
> >> My onreply_route is here:
> >>
> >> onreply_route[1] {
> >>
> >>
> >>
> >>         if (isflagset(2) && status =~ "(183)|2[0-9][0-9]") {
> >>
> >>
> >>
> >>                 fix_nated_contact();
> >>
> >>
> >>
> >>                 xlog("L_ERR", "%mb");
> >>                 force_rtp_proxy();
> >>
> >>
> >>
> >>         } else if (nat_uac_test("1")) {
> >>
> >>
> >>
> >>                 fix_nated_contact();
> >>         };
> >> }
> >>
> >>
> >>
> >>  0(27011) SIP/2.0 200 OK
> >> Via: SIP/2.0/UDP 67.184.42.101;branch=z9hG4bKe7ac.628388e2.0
> >> Via: SIP/2.0/UDP
> >> 192.168.0.83;rport=5060;received=68.184.42.171;branch=z9hG4bKb4bfc6084f306712
> >> From:
> >> <sip:9990010001 at sip.mycompany.com;user=phone>;tag=0d5452e4bee210e3
> >> To: "Andrew"
> >> <sip:9990010002 at sip.mycompany.com;user=phone>;tag=b73c75ac247b63db
> >> Call-ID: 0d5bc1fb337cd7eb at 68.184.40.199 CSeq: 26976 BYE User-Agent:
> >> Grandstream BT100 1.0.5.11 Contact:
> >> <sip:9990010002 at 68.184.40.199;user=phone> Allow:
> >> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
> >> Content-Length: 0
> >>
> >>
> >>
> >>
> >>
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> >>
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> >>
> >
> >
> >
> >
> > __________________________________
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> 
> 



		
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