[SR-Users] rr problem: sips instead of sip

[Daniel-Constantin Mierla]

Hello,

I think the rule is that if the r-uri is enforcing sips, then record
route has to be sips.

In general, sips is a mess, because the rule is: it has to be
transmitted over something secure (which can be udp via vpn, or udp in a
private network, etc.).

You can probably fix it in the config. After loose_route(), the next hop
is $du or $ru -- $du has priority if it is set, so you can try something
like:

if($du!=$null) {
    if($du=~"^sips:") {
         $du = "sip:" + $(du{s.subst,5,0});
    }
}

Alternative, for the initial invite, if it is sips in r-uri, then back
it in a variable, store some dummy sip uri, do record_route() and put
back the URI.

As an update in the code, it may be useful to add a mod param to control
if sips should be ignored or not.

Cheers,
Daniel

On 07/11/14 18:02, Jon Bonilla (Manwe) wrote:
> Hi all
>
> I'm experiencing a problem while using a mobile client on a NGCP mr3.5.1 (based
> on kamailio 4.1.6)
>
> The problem is as follows:
>
> A custom mobile client
> B blink for linux
>
> The path of a call from B to A would be this one:
>
> B --> Kamailio LB --> Kamailio proxy --> Sems --> Kamailio LB --> A
>
>
> Kamailio LB is a stateless proxy. Kamailio proxy is a stateful proxy and
> registrar and sems acts as B2BUA.
>
> Kamailio LB listens tcp,udp and tls to the outside and speaks udp to the
> inside.
> Kamailio proxy and sems only listen udp localhost (ports 5062 ans 5080).
>   
>
> My problem comes when the ACK from B to A (to the INVITE-200) doesn't leave the
> Kamailio LB to the proxy. The problem is that the kamailio proxy added itself
> as sips in the record-route header:
>
>
> U 2014/11/06 19:22:20.447717 127.0.0.1:5062 -> 127.0.0.1:5080
>
> INVITE sips:test1 at 10.0.1.5:41956;rinstance=145658D8;transport=tcp SIP/2.0'
>
> Record-Route:
> <sips:127.0.0.1:5062;lr=on;ftag=3Kwc9Aj.elAwdwWtB5yAXqdxR8rsyWi3;did=3e6.48e1;mpd=ii;ice_caller=strip;ice_callee=strip;aset=50;rtpprx=yes;vsf=aHRBd2tQQ3VhVkZGamdrN2lhanhodEF3aw-->'
>
> Record-Route:
> <sip:127.0.0.1;r2=on;lr=on;ftag=3Kwc9Aj.elAwdwWtB5yAXqdxR8rsyWi3;nat=yes;ngcplb=yes;socket=udp:XX.XX.XX.XX:5060>'
>
> Record-Route:
> <sip:XX.XX.XX.XX;r2=on;lr=on;ftag=3Kwc9Aj.elAwdwWtB5yAXqdxR8rsyWi3;nat=yes;ngcplb=yes;socket=udp:XX.XX.XX.XX:5060>'
>
> Via: SIP/2.0/UDP
> 127.0.0.1:5062;branch=z9hG4bKc36a.b42be7e4a4db27ac699a52fda7fb5fe0.0' Route:
> <sip:lb at 127.0.0.1;lr;received=sip:YY.YY.YY.YY:41956%3Btransport%3Dtls;socket=sip:XX.XX.XX.XX:5061>'
>
> Via: SIP/2.0/UDP
> 127.0.0.1;branch=z9hG4bKc36a.a58b9955ce93726b17c0240a555011f2.0' 
>
> Via: SIP/2.0/UDP
> 10.10.10.4:47879;received=ZZ.ZZ.ZZ.ZZ;rport=47879;branch=z9hG4bKPj6UmIovzyxDRCgcuz-VoXHEyJHRpA712x'
>
>
> When the ACK arrives to the LB from B, it can't relay it to the Kamailio proxy:
>
> NOTICE: <script>: New request on lb - M=ACK
> R=sip:ngcp-lb at XX.XX.XX.XX;ngcpct=7369703a3132372e302e302e313a353038303b707278726f7574653d31
>
> NOTICE: <script>: Relaying request,
> du='sips:127.0.0.1:5062;lr;ftag=3Kwc9Aj.elAwdwWtB5yAXqdxR8rsyWi3;did=3e6.48e1;mpd=ii;ice_caller=strip;ice_callee=strip;aset=50;rtpprx=yes;vsf=aHRBd2tQQ3VhVkZGamdrN2lhanhodEF3aw--',
> fs='udp:127.0.0.1:5060' - R=sip:127.0.0.1:5080;prxroute=1
>  
> WARNING:
> <core> [forward.c:264]: get_send_socket2(): WARNING: get_send_socket:
> protocol/port mismatch (forced udp:127.0.0.1:5060, to tls:127.0.0.1:5062)
>
>
>
>
> I think the problem might be in the way the client registers? This is the
> registration info where it uses sips in the contact:
>
> contact: sips:test1 at 10.0.1.5:41956;rinstance=145658D8;transport=tcp
>
>
> If I try the same calling blink, which has the following contact in the
> registration it works without issues as the proxy doesn't insert sips in the
> Record-Route header:
>
> contact: sip:91807432 at 10.10.10.4:57160;transport=tls
>
>
>
> Any thoughts? 
>
>
> cheers,
>
> Jon
>
>
>
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-- 
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Nov 24-27, Berlin - http://www.asipto.com