For instance, this week I got very confused over use of the term "cic" in telco-speak. I had thought cic was "circuit identification code". So, when one of our partners asked me to support embedding the cic in the incoming SIP INVITE, I freaked out. Why is that my responsibility? Why should my proxy ("class 5 switch") have to maintain state of individual DS0s on YOUR media gateways. This sucks! I bitterly started reading RFC 3398 (particularly section 7.2.1.1) and gave up halfway through it, becoming frustrated with the overall language of the section. After all, keeping track of DS0s just isn't my thing, and that's what CICs are about, right?
Turns out I was wrong. Keeping tracks of CICs is my thing. The problem is, these CICs are "Carrier Identification Codes", not "Circuit Identification Codes". I know everyone in networking and telecom loves to have an acronym for everything, but for GODs SAKE PEOPLE don't use the same acronym in the same context. Reading on in RFC 3398, they specifically warn for this confusion:
..
If the 'cic=' parameter is present in the Request-URI, the
gateway SHOULD consult local policy to make sure that it is
appropriate to transmit this Carrier Identification Code (CIC, not to
be confused with the MTP3 'circuit identification code') in the IAM;
..
Good point. Why didn't you say something sooner? So, now it turns out all I have to do is append a cic= (CARRIER identification code) parameter to the SIP RURI (request uniform resource identifier) on the outbound INVITE to the SIP <-> SS7 gateway. No problem. Thanks to OpenSER, module textops, and subst_uri, this is all I need to do:
avp_printf("$avp(s:cic)", "+15062"); # Set the cic value in an AVP
if (subst_uri('/^sip:([0-9]+)@(.*)$/sip:\1;cic=$avp(s:cic)@\2/i')) {
xlog("Added CIC $avp(s:cic) to RURI"); # tell us about it
};
Here is the INVITE going out to the gateway (cic in red):
U a.b.c.d:5060 -> w.x.y.z:5060
INVITE sip:14145551212;cic=+15062@w.x.y.z:5060 SIP/2.0.
Record-Route:
Via: SIP/2.0/UDP a.b.c.d;branch=z9hG4bKf1e2.8a4d41b6.0.
Via: SIP/2.0/UDP e.f.g.h:5060;branch=z9hG4bK4e28b419;rport=5060.
From: "Polycom 320"
To:
Contact:
Call-ID: 343b661709e271c30bdfafe923a82adb@e.f.g.h.
CSeq: 103 INVITE.
User-Agent: Star2Star StarBox astlinux-s2s-1438-net4801.
Max-Forwards: 69.
Remote-Party-ID: "Polycom 320"
Date: Fri, 22 Feb 2008 19:48:44 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Content-Type: application/sdp.
Content-Length: 265.
.
v=0.
o=root 6183 6184 IN IP4 e.f.g.h.
s=session.
c=IN IP4 e.f.g.h.
t=0 0.
m=audio 19770 RTP/AVP 18 0 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
Yeah, yeah, you like that?!?
I'm still not sure of the actual CIC format... That isn't EXACTLY what is shown in section 8.2.1.1 of RFC 3398 but it does look like what Cisco expects from their docs I've read (from the BTS 10200). After all, this particular SIP <-> SS7 gateway is made by them (PGW 2200).
Obviously you can get fancy with that and do all sorts of conditionals and DB calls to get the CIC, only append it on certain calls, different CICs for different sources, destinations, etc but the point is you can tweak up the RURI and insert it. I still haven't tested this against Cisco but I'm pretty sure it will work...
After I sorted out my "CIC confusion" it took me less than 30 minutes to implement this. Thanks OpenSER. You make SS7 a lot easier to deal with (as long as SIP is involved)!
No comments:
Post a Comment