[Asterisk-Dev] SIP REINVITE
Dan Evans
devans at invores.com
Mon May 16 06:08:20 MST 2005
Although the attached does not specifically address the problem below,
it is a summary of a protocol exchange which has a number of
questionable (re)INVITE's The exchange originates from a VoicePulse
call to an Asterisk system acting as a SIP proxy to an IVR. For some
reason, Voicepulse always sends two calls, then accepts the first one
answered and cancels the other one. But the issue in the trace is the
reINVITE's at lines 14, 18, 23, and 25. The first question is why are
they there, but if one looks at the SDP, the INVITE's appear to be eiher
sent in the wrong direction or contain the wrong c= elements. I have
the full raw trace that I can send off-list if someone is interested.
Dan
Olle E. Johansson wrote:
> BJ Weschke wrote:
>
>> Server A (IP 192.168.1.1)
>> Server B (IP(s) 192.168.1.2 [actual] 192.168.1.3 [vip])
>> Server C (IP(s) 192.168.1.4)
>>
>> All servers are Asterisk installs. All servers have SIP canreinvite=yes.
>>
>> Server A calls Server B on his VIP. The call sets up fine, but the
>>3rd of 4th step in the dial plan is to then transfer that call on to
>>Server C. Server B dials server C and then begins to attempt a native
>>bridge between Server A and C. Server A responds back with "SIP/2.0
>>482 Loop Detected" assumably because the man in the middle has
>>different terminating/originating IP addresses and has sent an
>>improper invite back to A to start the briding process.
>
> Can you send me a packet trace of this?
>
>
>> Does ANTHM's patch from a few weeks back to chan_sip fix this
>>problem, or is this still a "live" issue? If it is patched, who needs
>>the patch in the scenario above? Just server B? or Servers A and C
>>too?
>
> I haven't seen the loop detected issue, but understand where it's coming
> from. Anthm's patch is more to be seen as a proof-of-concept than
> something you want to use. I'm trying to continue the work based on his
> patch, but it will require a lot of changes to chan_sip.
>
> I'm glad to see another person wanting to transfer calls from Asterisk
> to another SIP domain - I just had a question from a core developer on
> the theme "why would anyone want to do that?"... So I needed your mail
> to prove that's it is not only me and my customers that need that function.
>
> Digging into how chan_sip handles transfers I'm amazed that it work with
> anything... ;-)
>
> /Olle
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-------------- next part --------------
VoicePulse Asterisk IVR
1 ---INVITE 266fc3ef ---------->| | |
2 |<--100 TRYING 266fc3ef -------| |
3 |---INVITE 124eb4d1 ---------->| |
4 ---INVITE 47e70fca ---------->| | |
5 |<--100 TRYING 47e70fca -------| |
6 |---INVITE 2de13285 ---------->| |
7 | |<--180 RINGING 124eb4d1 ------|
8 |<--180 RINGING 266fc3ef ------| |
9 | |<--180 RINGING 2de13285 ------|
10 |<--180 RINGING 47e70fca ------| |
11 | |<--200 OK 124eb4d1 -----------|
12 |---ACK 124eb4d1 ---------->| |
13 |<--200 OK 266fc3ef ------| |
14 |---INVITE 124eb4d1 ---------->| |
15 | |<--200 OK 2de13285 -----------|
16 |---ACK 2de13285 ---------->| |
17 |<--200 OK 47e70fca ------| |
18 |---INVITE 2de13285 ---------->| |
19 ---CANCEL 47e70fca ---------->| | |
20 |<--487 Req Term 4e70fca ------| |
21 |<--200 OK (cancel) 47e70fca --| |
22 ---ACK 266fc3ef ---------->| | |
23 |<--INVITE 266fc3ef ------| |
24 ---ACK 47e70fca ---------->| | |
25 |<--INVITE 47e70fca ------| |
26 ---ACK 47e70fca ---------->| | |
27 ---200 OK 266fc3ef ---------->| | |
28 |<--ACK 266fc3ef ------| |
29 ---404 Not Found 47e70fca --->| | |
30 |<--ACK 47e70fca ------| |
31 | |<--200 OK 124eb4d1 -----------|
32 |---ACK 124eb4d1 ---------->| |
33 | |<--200 OK 2de13285 -----------|
34 |---ACK 2de13285 ---------->| |
35 |---BYE 2de13285 ---------->| |
36 | |<--200 OK 2de13285 -----------|
37 ---BYE 266fc3ef ---------->| | |
38 |<--200 OK 266fc3ef ------| |
39 |---INVITE 124eb4d1 ---------->| |
40 | |<--200 OK 124eb4d1 -----------|
41 |---ACK 124eb4d1 ---------->| |
42 |---BYE 124eb4d1 ---------->| |
43 | |<--200 OK 124eb4d1 -----------|
44
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