[OpenSIPS-Users] Opensips + rtpproxy + SBC
Raistlin Majere
raistmaj at gmail.com
Wed Sep 30 17:05:56 CEST 2015
First of all thanks for your quick response.
At the moment this is the point where we have the t_on_reply for that
onreply_route I sent in the previous email.
route[1] {
if(!cache_fetch("local", "outerip_$var(destinationIp)", $avp(outerIp))) {
get_outer_ip("$var(destinationIp)", "$avp(outerIp)");
cache_store("local", "outerip_$var(destinationIp)", "$avp(outerIp)", 10);
}
$fs = $proto + ":" + $avp(outerIp);
if (is_method("BYE|CANCEL")) {
unforce_rtp_proxy();
} else if (is_method("INVITE")){
if (has_body("application/sdp")) {
$var(trustconnectionip) = "%TRUSTCONNECTIONIP%";
$var(ciptrusted) = "no";
if ($var(trustconnectionip)=="yes") {
$var(ciptrusted) = "yes";
} else if ($var(trustconnectionip)=="auto") {
$var(sdpc) = $(rb{sdp.line,c}{s.substr,9,0});
if($td == $fd && $td != $var(sdpc)) {
$var(ciptrusted) = "yes";
}
}
if ($var(ciptrusted)=="yes") {
rtpproxy_offer("focnr");
} else {
rtpproxy_offer("focn");
}
}
}
# Prevent $var(destinationPort) from getting default port number 5060 for
TLS if no port specified in R-URI.
if($(ru{uri.port})=="" && $proto=="tls" && $var(destinationPort)=="5060") {
$var(destinationPort) = 5061;
}
# force the transport protocol to the same one the client used
$du =
"sip:"+$var(destinationIp)+":"+$var(destinationPort)+";transport="+$proto;
t_on_reply("1");
# for INVITEs enable some additional helper routes
if (is_method("INVITE")) {
t_on_failure("1");
}
if (!t_relay()) {
xlog("L_INFO", "Relay error");
sl_reply_error();
}
exit;
}
The route[1] is set in the main route under this conditions
if (has_totag()) {
# sequential request within a dialog should
# take the path determined by record-routing
if (loose_route()) {
$var(destinationIp) = $avp(requestIp);
$var(destinationPort) = $rp;
if (is_method("INVITE")) {
# even if in most of the cases is useless, do RR for
# re-INVITEs alos, as some buggy clients do change route set
# during the dialog.
record_route();
}
# route it out to whatever destination was set by loose_route()
# in $du (destination URI).
route(1);
...
exit;
}
....
And if there is no REGISTER, PUBLISH, CANCEL
route(1);
}
About the engage_rtp_proxy I was actually thinking about it but I'm a bit
scared because of the bridge mode, as it is a "generic" script, I need to
identify all this cases and create configuration cases to make scripts
automatically for those setups.
Srly, thanks for the response >.< I have only 1 week experience with this
protocol and I'm having nightmares already.
Kind regards
Jose Palma
2015-09-30 16:51 GMT+02:00 Trevor Steyn <trevor at webon.co.za>:
> HI Raistlin,
>
> from your script we cannot see where you are setting t_on_reply routes as
> im sure this is where you are probably missing the rtpproxy_answer() but
> cannot see why as that part of the script is missing
>
>
> maybe you can also try use engage_rtp_proxy() and you wont have to worry
> about the answer and you could remove the reply routes and just relay
>
> if ($var(ciptrusted)=="yes") {
> engage_rtp_proxy("focnr");
> } else {
> engage_rtpproxy("focn");
> }
>
> Regards
> Trevor Steyn
>
>
>
>
>
> On 30/09/2015 14:56, Raistlin Majere wrote:
>
> Hi,
>
> Recently the maintainer of the SIPs proxy in our company quit, and well
> I'm the new in charge of this project, the bad new is I had 0 experience
> with SIP. After some week I got my first case related to our "SIP proxy".
>
> We are using OpenSIP 1.8 within our Firewall to handle the protocol and
> the NAT that it will imply within a Firewall. The script my ex coworker did
> is working in 99% of cases but this specific case.
>
> The customer has one setup like this
>
>
> PhoneA
> PhoneB
> PhoneC --- Call Manager --- Firewall --- SBC --- Farm of RTP Media servers
> ....
> PhoneN
>
> The opensips instance is running within the firewall. The next IPs are
> fake but follow the "rules" of internal/external it is just to avoid
> problems
>
> Call Manager: 172.17.1.1
> Firewall: Internal Network 192.168.0.10
> Firewall: Extenal Network 62.1.1.10
> SBC: 210.200.100.100
> Farm of Media Servers: 210.200.100.128/25
>
> So the invite works as Expected but on the 180 Ringing either 200 OK the
> moment the messages traverse the SIP proxy, doesn't contain the "farm" IP
> but the SBC IP.
>
> This are the 180 Ringing:
>
> From SBC to the Firewall
>
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 62.1.1.10:5060;branch=z9hG4bKd091.937a047.0
> Via: SIP/2.0/UDP 172.17.1.1:5060
> ;rport=5060;received=172.17.1.1;branch=z9hG4bKac393424402
> From: <sip:5000 at 210.200.100.100>;tag=1c393411873
> To: <sip:5001 at 210.200.100.100;user=phone>;tag=gK08c71cc5
> Call-ID: 39341083229920151062 at 172.17.1.1
> CSeq: 1 INVITE
> Record-Route: <sip:62.1.1.10:5060;r2=on;lr;did=6d8.933abaa6>
> Record-Route: <sip:192.168.0.10:5060;r2=on;lr;did=6d8.933abaa6>
> Contact: <sip:5001 at 210.200.100.100:5060>
> Allow:
> INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
> Require: 100rel
> RSeq: 433990
> Content-Length: 266
> Content-Disposition: session; handling=required
> Content-Type: application/sdp
>
> v=0
> o=Sonus_UAC 176482 50736 IN IP4 210.200.100.100
> s=SIP Media Capabilities
> c=IN IP4 210.200.100.243
> t=0 0
> m=audio 61348 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendrecv
> a=ptime:20
> a=silenceSupp:off - - - -
>
> After the firewall + Opensips have processed this message to the call
> center
>
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 172.17.1.1:5060;branch=z9hG4bKac393424402
> From: <sip:5000 at 210.200.100.100>;tag=1c393411873
> To: <sip:5001 at 210.200.100.100;user=phone>;tag=gK08c71cc5
> Call-ID: 39341083229920151062 at 172.17.1.1
> CSeq: 1 INVITE
> Record-Route: <sip::62.1.1.10:5060;r2=on;lr;did=6d8.933abaa6>
> Record-Route: <sip:192.168.0.10:5060;r2=on;lr;did=6d8.933abaa6>
> Contact: <sip:5001 at 210.200.100.100>
> Allow:
> INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
> Require: 100rel
> RSeq: 433990
> Content-Length: 295
> Content-Disposition: session; handling=required
> Content-Type: application/sdp
>
> v=0
> o=Sonus_UAC 176482 50736 IN IP4 210.200.100.100
> s=SIP Media Capabilities
> t=0 0
> m=audio 4845 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendrecv
> a=ptime:20
> a=silenceSupp:off - - - -
> a=nortpproxy:yes
> c=IN IP4 210.200.100.100
> a=rtcp:4848
>
> ----
>
> The RTP "acceptor" is created under the IP 210.200.100.100 instead of the
> IP 210.200.243 as the SIPproxy is changing the SDP connection information.
>
>
> This is the logic we are using in our script for the INVITE and for the
> onreply_route
>
> if (is_method("INVITE")){
> if (has_body("application/sdp")) {
> $var(trustconnectionip) = "%TRUSTCONNECTIONIP%";
> $var(ciptrusted) = "no";
> if ($var(trustconnectionip)=="yes") {
> $var(ciptrusted) = "yes";
> } else if ($var(trustconnectionip)=="auto") {
> $var(sdpc) = $(rb{sdp.line,c}{s.substr,9,0});
> if($td == $fd && $td != $var(sdpc)) {
> $var(ciptrusted) = "yes";
> }
> }
> if ($var(ciptrusted)=="yes") {
> rtpproxy_offer("focnr");
> } else {
> rtpproxy_offer("focn");
> }
> }
> }
>
>
>
> And on the onreply
>
> if (has_body("application/sdp")) { $var(trustconnectionip) =
> "%TRUSTCONNECTIONIP%"; $var(ciptrusted) = "no"; if
> ($var(trustconnectionip)=="yes") { $var(ciptrusted) = "yes"; } else if
> ($var(trustconnectionip)=="auto") { $var(sdpc) =
> $(rb{sdp.line,c}{s.substr,9,0}); if($td == $fd && $td != $var(sdpc)) {
> $var(ciptrusted) = "yes"; } } if ($var(ciptrusted)=="yes") {
> rtpproxy_answer("fr"); } else { rtpproxy_answer("f"); } }
>
>
> Where TRUSTONNECTIONIP = "no" so basically we are doing
>
> rptproxy_offer("focn") and rtpproxy_answer("f").
>
> Kind regards:
>
> Jose Palma
>
>
>
>
> _______________________________________________
> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20150930/4b98d057/attachment-0001.htm>
More information about the Users
mailing list