Any substitute or replace function for the request-/response line of SIP messages available?
Hi,
I have a very specific situation that the implementation of a customer SIP user agent is absolutely not standards conform. Therefore, I use Kamailio as a SIP protocol adapter between the non-standard conform user agent(s) and the typical RFC3261 compliant user agents. Most conversions are working fine. But in a very specific scenario I have to change/convert a SIP response message to a SIP request message and change the SIP method´s name. This conversion is no problem for the SIP header fields. However, I haven't found a way for adapting the response-/request line of a SIP message. This line seems to be protected.
The described situation of the perverse conversion I try to realize is following (please do not ask "why?" or think I'm mad...): (mad) UA1 == INVITE ==> SIP-Prot-Adapter(Kamailio) == INVITE ==> (typ.)UA2 UA1 <== BYE == SIP-Prot-Adapter(Kamailio) <== 488 == (typ.)UA2 UA1 == BYE ==> SIP-Prot-Adapter(Kamailio) == ACK ==> (typ.)UA2
I use the "textops" module for substitution and/or replacement in SIP header fields. But the "subst" and "replace" functions cannot be used for the request-/response line on top of the SIP messages. I was a little bit irritated, as the "search" function is still working on these lines, but not the subst/replace function. Creating a new SIP message with the UAC module (as alternative) is not okay, as this new message does not use the same values for Call-ID, From-tag a.s.o. and these values can't be changed before sending the message.
Does anybody know a way / function / method for manipulating the request-/response line of a message? It would be perfect using a function á la "sed"/subst.
Thanks Klaus
P.S. I tested with kamailio-3.3.3
At the risk of sounding unhelpful, if you are having to modify the request/status line, there's probably something more deeply wrong here, and you should probably endeavour to fix the problem in a somewhat more realistic manner.
Furthermore, it must be said that proxies do not make a particularly good "protocol adaptor"; in contrast to "thicker" back-to-back user agents (B2BUAs), which build out two logically independent call legs and bridge signaling events between them selectively, as they see fit, proxies are more or less obligated, in principle, to pass on what they receive, in unadulterated form. Kamailio does offer some hacks around that, in the form of textops and so on, but it doesn't really mean that it is the right tool for the job.
That said, if you are dead-set on doing this, your best for regenerating a reply into a request is the 'uac' module and uac_req_send() functionality:
http://kamailio.org/docs/modules/3.3.x/modules_k/uac.html#id2494432
-- Alex
On 02/15/2013 03:22 PM, Klaus Feichtinger wrote:
I know that it is an untypical way and that a B2BUA would be the best solution - that is absolutely clear. But because of a missing framework and time I decided to try it with Kamailio that I've used in many other projects, too.
Textops is still used and working fine for any headers (except request line). But UAC (as you´ve recommended) didn't satisfy me. I missed a possibility for setting dialog-specific settings á la Call-ID, from-tag, to-tag and CSeq. When I´ve tried setting the CSeq header with the function
$uac_req(hdrs)="CSeq: 123 INVITEr\n";
it has only added a second CSeq header to the outgoing message, but didn't replace the original CSeq header that was created by the UAC module. Is there a way to influence the listed id´s, too?
Klaus
...I think in worst case a controlled local loop (which was helpful in other situations, too), could be a solution for fixing the UAC problem.
1. UAC (module) sending the created request to the SIP server again (loop) 2. the server is doing textops and 3. is finally forwarding the request to the real target
I will try that.
Klaus
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Alex Balashov -
Klaus Feichtinger