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

[Henning Westerholt]

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

-- 
Henning Westerholt
https://skalatan.de/blog/