[SR-Users] bad transport for sips uri?

[David Cunningham]

Hi Daniel,

Certainly, here is the sequence including INVITE and 200 OK. I note that
the Contact on the 200 OK doesn't mention TLS, but the Record-Route header
does.
Thank you.

Session Initiation Protocol (INVITE)
    Request-Line: INVITE sips:105 at es8.example.com:5061 SIP/2.0
    Message Header
        Via: SIP/2.0/TLS 50.78.xx.xx:5061;branch=z9hG4bK1836763299
        Route: <sip:70.42.yy.yy:5061;transport=tls;lr>
        From: ES8 Test 102 <sips:111111 at es8.example.com
>;tag=Nf5GG!Orq!MSmGfeu66F03F2114df82b
        To: <sips:105 at es8.example.com:5061>
        Call-ID: 222222 at 50.78.xx.xx
        CSeq: 94679 INVITE
        Contact: <sips:111111 at 50.78.xx.xx:5061;transport=tls>
        Supported: 100rel
        Proxy-Authorization: Digest username="111111", realm="
es8.example.com", nonce="XG71VFxu9CjpZnZTIQhhf5V1KmKWmQes", uri="
sip:105 at 192.168.3.1;user=phone",
response="60059b561416f1a184d556d5cf116014", algorithm=MD5
        Max-Forwards: 70
        User-Agent: ewb2bua/15.3.0
        Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO,
PRACK, UPDATE, MESSAGE
        Content-Type: application/sdp
        Content-Length:   387
    Message Body


Session Initiation Protocol (200)
    Status-Line: SIP/2.0 200 OK
    Message Header
        Via: SIP/2.0/TLS
50.78.xx.xx:5061;rport=43290;branch=z9hG4bK1836763299
        Record-Route: <sips:70.42.yy.yy;r2=on;lr=on>
        Record-Route: <sips:70.42.yy.yy:5061;transport=tls;r2=on;lr=on>
        From: ES8 Test 102 <sips:111111 at es8.example.com
>;tag=Nf5GG!Orq!MSmGfeu66F03F2114df82b
        To: <sips:105 at es8.example.com:5061>;tag=as7a72b209
        Call-ID: 222222 at 50.78.xx.xx
        CSeq: 94679 INVITE
        Server: ES8
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
        Supported: replaces, timer
        Contact: <sips:105 at 70.42.yy.yy:5070>
        Content-Type: application/sdp
        Content-Length: 337
    Message Body


Session Initiation Protocol (ACK)
    Request-Line: ACK sips:105 at 70.42.xx.xx:5070 SIP/2.0
    Message Header
        Via: SIP/2.0/TLS 50.78.yy.yy:5061;branch=z9hG4bK741961203
        Route: <sips:70.42.xx.xx:5061;transport=tls;r2=on;lr=on>
        Route: <sips:70.42.xx.xx;r2=on;lr=on>
        From: ES8 Test 102 <sips:111111 at es8.example.com
>;tag=Nf5GG!Orq!MSmGfeu66F03F2114df82b
        To: <sips:105 at es8.example.com:5061>;tag=as7a72b209
        Call-ID: 222222 at 50.78.yy.yy
        CSeq: 94679 ACK
        Contact: <sips:111111 at 50.78.yy.yy:5061;transport=tls>
        Max-Forwards: 70
        User-Agent: ewb2bua/15.3.0
        Content-Length: 0


On Fri, 22 Feb 2019 at 20:16, Daniel-Constantin Mierla <miconda at gmail.com>
wrote:

> Hello,
>
> do you have the INVITE and the 200ok corresponding to this ACK? I need to
> see the R-URI of INVITE as well as the contact in 200ok in order to analyze
> why sips appears in ACK.
>
> SIPS requirements are sort of a mess, probably the best would be to let it
> for config to decide when to allow forwarding or not in case of sips URI.
>
> Cheers,
> Daniel
> On 22.02.19 02:58, David Cunningham wrote:
>
> Hello all,
>
> We're having an issue with Kamailio not processing an ACK and hope someone
> can help. A PCAP shows that Kamailio is receiving the ACK, and we believe
> these log messages are directly related to it:
>
> Feb 21 10:54:01 hostname /sbin/kamailio[15854]: ERROR: tm [ut.h:279]:
> uri2dst2(): ERROR: uri2dst: bad transport for sips uri: 1
> Feb 21 10:54:01 hostname /sbin/kamailio[15854]: ERROR: tm [t_fwd.c:1777]:
> t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches
> Feb 21 10:54:01 hostname /sbin/kamailio[15854]: ERROR: sl
> [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used:
> Unresolvable destination (478/SL)
>
> The ACK is as follows. It's from telephone 111111 which is using TLS,
> which Kamailio listens for on port 5061.
> Does anyone know which URI has the bad transport as per the error above?
> Might it be the From URI "sips:111111 at es8.example.com" because it doesn't
> specify port 5061?
> Thank you in advance.
>
> Session Initiation Protocol (ACK)
>     Request-Line: ACK sips:105 at 70.42.xx.xx:5070 SIP/2.0
>     Message Header
>         Via: SIP/2.0/TLS 50.78.yy.yy:5061;branch=z9hG4bK741961203
>         Route: <sips:70.42.xx.xx:5061;transport=tls;r2=on;lr=on>
>         Route: <sips:70.42.xx.xx;r2=on;lr=on>
>         From: ES8 Test 102 <sips:111111 at es8.example.com
> >;tag=Nf5GG!Orq!MSmGfeu66F03F2114df82b
>         To: <sips:105 at es8.example.com:5061>;tag=as7a72b209
>         Call-ID: 222222 at 50.78.yy.yy
>         CSeq: 94679 ACK
>         Contact: <sips:111111 at 50.78.yy.yy:5061;transport=tls>
> <sips:111111 at 50.78.yy.yy:5061;transport=tls>
>         Max-Forwards: 70
>         User-Agent: ewb2bua/15.3.0
>         Content-Length: 0
>
> --
> David Cunningham, Voisonics Limited
> http://voisonics.com/
> USA: +1 213 221 1092
> New Zealand: +64 (0)28 2558 3782
>
> _______________________________________________
> Kamailio (SER) - Users Mailing Listsr-users at lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
> --
> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda
> Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com
> Kamailio Advanced Training - Mar 4-6, 2019 in Berlin; Mar 25-27, 2019, in Washington, DC, USA -- www.asipto.com
>
>

-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
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