[OpenSIPS-Users] Broken Loose Routing?
Brett Nemeroff
brett at nemeroff.com
Wed Nov 5 15:30:27 CET 2008
No, I'm not. I'm not sure what to do to completely remove the contact. Am I
doing something wrong?
In a related problem... if I use remove_hf and insert_hf, it leaves the
partial header without the "Contact" bit and angle brackets, and inserts a
properly formatted Contact header. The call still completes. However, the
200 OK that comes back never seems to "work". The call never sets up. I'm
not sure there is any point in troubleshooting without the headers being
right. Right now I'm only doing this fix on INVITEs coming from the inside
network.
On Wed, Nov 5, 2008 at 2:24 AM, Bogdan-Andrei Iancu
<bogdan at voice-system.ro>wrote:
> Hi Brett,
>
> are you doing any other changes on Contact hdr? like calling
> fix_nated_contact() or other function that might change the contact?
>
> Regards,
> Bogdan
>
> Brett Nemeroff wrote:
>
>> Ok, I feel really silly. I've tried a dozen ways to rewrite the contact
>> header.. Over and over it doesn't work. First I tried to replace the contact
>> header (which is how I wanted to do it):
>> subst('/^Contact: <sip:([0-9]+)@(.*)$/Contact: <sip:\1 at 64.174.184.141:5060
>> ;maddr=\2>/ig');
>>
>> That mostly worked, but produced a header that looks like this:
>> Contact: <sip:5122033884 at 64.174.184.141:5060;maddr=10.10.67.190 <
>> http://10.10.67.190>>sip:5125551212 at 10.10.67.190:5060 <
>> http://sip:5125551212@10.10.67.190:5060>
>>
>> In other words, it removed the angle braces from the original header, and
>> put the substituted text before it. But it didn't remove the header, which
>> matches the pattern
>>
>> I've even tried doing: subst('/^Contact:.*$//');
>> and
>> remove_hf("Contact");
>>
>> Both just removed the header field, but left the contents, WITHOUT the
>> angle braces (see the line without a header FIELD):
>> Call-ID: 19380521914112008222455 at 10.10.67.190 <mailto:
>> 19380521914112008222455 at 10.10.67.190>.
>> CSeq: 1 INVITE.
>> sip:5125551212 at 10.10.67.190:5060 <http://sip:5125551212@10.10.67.190:5060
>> >
>> Supported: em,100rel,timer,replaces,path,early-session,resource-priority.
>> Allow:
>> REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE.
>>
>> I'm sure I must be doing something wrong. Is there a "right way" to change
>> the contact header? Perhaps it's not something I can change with the
>> standard toolset?
>>
>> -Brett
>>
>> On Thu, Oct 30, 2008 at 10:22 AM, Bogdan-Andrei Iancu <
>> bogdan at voice-system.ro <mailto:bogdan at voice-system.ro>> wrote:
>>
>> Hi Brett,
>>
>> Not sure you can do anything about - they detect you as behind nat
>> because of the private IP in Contact....
>>
>> What you can do is to try (before sending the INVITE to the
>> provider) to save the current contact IP in a uri param and to
>> replace the contact IP with your own....Something like maddr
>>
>> When you get a sequential request , before loose_route(), replace
>> the RURI IP with the value from param.
>>
>> Something like:
>>
>> INVITE : Contact: <sip:user at privIP> -> Contact:
>> <sip:user at server_pub_IP;maddr=privIP>
>>
>> BYE: RURI= sip:user at server_pub_IP;maddr=privIP ->
>> RURI=sip:user at privIP
>>
>>
>> Regards,
>> Bogdan
>>
>> Brett Nemeroff wrote:
>>
>> We're actually getting this problem sending calls to multiple
>> providers.. is there possibly anything I'm dong that could be
>> causing this?
>> Thanks all,
>> Brett
>>
>>
>>
>> On Thu, Oct 30, 2008 at 11:08 AM, Alex Hermann
>> <alex at speakup.nl <mailto:alex at speakup.nl>
>> <mailto:alex at speakup.nl <mailto:alex at speakup.nl>>> wrote:
>>
>> On Thursday 30 October 2008, Iñaki Baz Castillo wrote:
>> > 2008/10/30 Brett Nemeroff <brett at nemeroff.com
>> <mailto:brett at nemeroff.com>
>> <mailto:brett at nemeroff.com <mailto:brett at nemeroff.com>>>:
>>
>>
>> > > Please note, the provider isn't sending the BYE to the
>> internal address.
>> > > The provider sends it to OpenSIPs Public address. Which is
>> correct..
>> >
>> > No, that's not correct, the RURI of the BYE from provider
>> *MUST* be
>> > the same as the RURI in the INVITE Contact the provider
>> received.
>> > The BYE is sent back to the public IP, of course, but
>> it's routed
>> > thanks to "Route" header (loose route).
>> >
>> > This is the INVITE OpenSIPS sends to the provider:
>> >
>> > U 62.175.182.141:5060 <http://62.175.182.141:5060>
>> <http://62.175.182.141:5060> ->
>> 195.62.163.131:5060 <http://195.62.163.131:5060>
>> <http://195.62.163.131:5060>
>>
>> > INVITE sip:17475551212 at 195.62.163.131:5060;transport=udp
>> SIP/2.0.
>> > Contact: <sip:15125551234 at 10.10.67.190:5060
>> <http://sip:15125551234@10.10.67.190:5060>
>> <http://sip:15125551234@10.10.67.190:5060>>.
>>
>> >
>> > And this is the BYE the provider sends to OpenSIPS:
>> >
>> > BYE sip:15125551234 at 62.175.182.141:5060
>> <http://sip:15125551234@62.175.182.141:5060>
>> <http://sip:15125551234@62.175.182.141:5060> SIP/2.0.
>>
>>
>> Most probably the provider does wrong NAT detection. ie it does
>> the equivalent
>> of:
>>
>> if (nat_uac_test("1")) {
>> fix_nated_contact();
>> }
>> t_relay();
>>
>>
>> --
>> Met vriendelijke groet,
>>
>>
>> Alex Hermann
>> SpeakUp
>> T: 088-SPEAKUP (088-7732587)
>> F: 088-7732588
>>
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