[SR-Users] uac_replace_from/to, msg_apply_changes and Re-INVITE

[Daniel-Constantin Mierla]

Hello,

trying to think a bit more about it, I guess that what happens there is
the following:

  - re-invite comes it
  - loose_route() is executed and by that uac module has the callback
for updating To/From executed
  - you update the sdp and then call apply changes, which updates also
the headers
  - transaction is created with the new values for From/To headers

If my assumption is right, then the solution would be to do the updates
and apply changes before calling loose_route().

Cheers,
Daniel

On 28.03.19 19:39, Henning Westerholt wrote:
> Hi Daniel,
>
> Thank you. That was a really good Idea, this workaround seems to work just fine.
>
> Best regards,
>
> Henning
>
> Am 28. März 2019 14:51:51 MEZ schrieb Daniel-Constantin Mierla <miconda at gmail.com>:
>> Hello,
>>
>> didn't get the chance to look at it yet, being out of the office and
>> the
>> little spare time I got these days I tried to look into
>> libssl/libcrypto
>> 1.1 for the blocking issue reported recently.
>>
>> I haven't run uac replace with dialog, but as you said, it should not
>> be
>> an impact of that.
>>
>> When you run with debug=3, couldn't you spot any hint about what can go
>> wrong there?
>>
>> As for a workaround until it gets fixed, try sending 100 trying with sl
>> module and set the flag for no  automatic 100 from tm.
>>
>> Cheers,
>> Daniel
>>
>> On 28.03.19 07:19, Henning Westerholt wrote:
>>> Hello,
>>>
>>> did you had a chance to have a quick look? Any sugestions are
>> welcome.
>>> Best regards,
>>>
>>> Henning
>>>
>>> Am 26. März 2019 08:13:20 MEZ schrieb Henning Westerholt
>> <hw at kamailio.org>:
>>>> Am Dienstag, 26. März 2019, 02:27:17 CET schrieb Daniel-Constantin
>>>> Mierla:
>>>>>> I am currently debugging a strange issue related to the usage of
>>>>>> uac_replace_from/to() after msg_apply_changes(). It works all
>> fine,
>>>> but in
>>>>>> a Re-INVITE case it inserts the wrong headers for the 100 - Trying
>>>> case.
>>>>>> The calle
>>>>> did you mean caller or callee? What side is sending the re-INVITE
>>>> with
>>>>> the troubles?
>>>>>
>>>>> Is uac using the record-route param to store the info about
>> From/To?
>>>> Or
>>>>> is via dialog variables?
>>>> Hi Daniel,
>>>>
>>>> storage is with dialog variables, but in the 100 - Trying case this
>> is 
>>>> internally done with the uac TM callbacks (If I understood the code 
>>>> correctly). If I remove the msg_apply_changes() in the Re-INVITE
>> case,
>>>> all 
>>>> works fine.
>>>>
>>>> Call flow (A is a SIP client, B a PSTN destination):
>>>>
>>>> A -> B INVITE, 100, 18x, 200, ACK: all fine
>>>> A -> B first Re-INVITE, 100: the 100 - Trying is broken
>>>> A -> B second Re-INVINTE, 100: the 100 - Trying is broken
>>>> ... more Re-INVINTEs 
>>>> A -> B BYE: all fine
>>>>
>>>> I have attached a trace, see below the 100 - Trying examples:
>>>>
>>>> This is ok:
>>>>
>>>>    Kamailio-IP.sip > A-SIDE-IP.na-localise: SIP, length: 405
>>>>        SIP/2.0 100 trying -- your call is important to us
>>>>        Via: SIP/2.0/UDP A-SIDE-IP:
>>>>
>> 5062;rport=5062;branch=z9hG4bKPjh2a4HyDf1ApI7kWK5ShgL7I7w9.LvoeI;received=A-
>>>> SIDE-IP
>>>>    From:
>> sip:A-Number at Kamailio-IP;tag=Ho4cXwV4Q-LBW0n4guPOQwvi6GbiylqO
>>>>        To: sip:B-Number at Kamailio-IP
>>>>        Call-ID: po-DL6LvwksnKK56D3lF8HB4wwo6F.GS
>>>>        CSeq: 6764 INVITE
>>>>        Server: Carrier-Name SIP Voice Gateway
>>>>        Content-Length: 0
>>>>
>>>> This is not OK:
>>>>
>>>> 12:42:59.336930 IP (tos 0x10, ttl 64, id 61398, offset 0, flags
>> [none],
>>>> proto 
>>>> UDP (17), length 461)
>>>>    Kamailio-IP.sip > A-SIDE-IP.na-localise: SIP, length: 433
>>>>        SIP/2.0 100 trying -- your call is important to us
>>>>  Via: SIP/2.0/UDP
>> A-SIDE-IP:5062;rport=5062;branch=z9hG4bKPj3jfO6Fuk5-
>>>> SJQ2EXOIEeYJk1amH.xKvp;received=A-SIDE-IP
>>>>  From:
>> sip:+Rewritten-Number at sip.voice.Carrier-Name.net;tag=Ho4cXwV4Q-
>>>> LBW0n4guPOQwvi6GbiylqO
>>>>        To: sip:+B-Number at Carrier-IP:5060;tag=7N2BS4K70tZ9Q
>>>>        Call-ID: po-DL6LvwksnKK56D3lF8HB4wwo6F.GS
>>>>        CSeq: 6765 INVITE
>>>>        Server: Carrier-Name SIP Voice Gateway
>>>>        Content-Length: 0
>>>>
>>>> Cheers,
>>>>
>>>> Henning

-- 
Daniel-Constantin Mierla -- www.asipto.com
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