Tag Archives: dial plan

CUBE URI-based Routing and Multiple Via Headers

Cisco introduced some pretty cool URI enhancements for CUBE from 15.4(1)T that have added some great extensions to the CUBE feature set, and specifically include some fine-grained SIP routing control.  These have been long overdue as IOS-based VoIP routing has been heavily dependent on numerical pattern matching due to Cisco’s ongoing support for H.323-style legacy configuration.

These enhancements are covered in a lovely Cisco doc here.  I’ve also blogged previously on a similar topic on how to utilize these and other new CUBE SIP features to provide Loop Prevention on CUBE.

 

To start, the basic configuration for URI dial-peer matching is somewhat trivial:

!
voice class uri voice-class-uri-tag sip|tel
    host hostname-pattern
    host ipv4:ipv4-address
    host ipv6:ipv6-address
    host dns:dns-address
    pattern uri-pattern
    user-id username-pattern
!
dial-peer voice tag voip
 session protocol sipv2
 incoming uri { from | request | to | via } voice-class-uri-tag
!

Here we see that incoming called-number is dropped and is instead routing using incoming uri.  Typically, we’d match the URI on a via header, with our voice class uri container applying a match on our preferred IP Address / DNS address.

I am referring to RFC compliant behaviour for the Via header from RFC 3261:

8.1.1.7 Via

   The Via header field indicates the transport used for the transaction
   and identifies the location where the response is to be sent.  A Via
   header field value is added only after the transport that will be
   used to reach the next hop has been selected (which may involve the
   usage of the procedures in [4]).

   When the UAC creates a request, it MUST insert a Via into that
   request.  The protocol name and protocol version in the header field
   MUST be SIP and 2.0, respectively.  The Via header field value MUST
   contain a branch parameter.  This parameter is used to identify the
   transaction created by that request.  This parameter is used by both
   the client and the server.

Here we note that a SIP Request can contain multiple Via headers.  This is used as a loop prevention mechanism for SIP Routing.  This becomes incredibly import when multiple SIP Proxies forward requests.  This is well described here for the SIP initiate.

 

With this in mind, I came across an interesting scenario recently where a call route through an ITSP’s SIP Proxy network changed without my knowledge, meaning that the Via header ordered list in the inbound INVITE changed.  This resulted in not matching my voice class uri configuration.  This stymied me at first until I noted the following for this feature:

In a scenario where multiple SIP hops are involved in a call, there would 
be multiple via headers involved, and the topmost via header of an 
incoming SIP invite represents the last hop that forwarded the SIP 
request, and the bottom-most via header would represent the originator 
of the SIP request. This feature supports matching by the last hop that 
forwarded the request (neighboringSIPentity), which is the topmost via 
header.

 

From the PDF link’s worked examples, we can see the matching will use the following behaviour:

INVITE sip:123@1.2.3.4:5060 SIP/2.0

Via: SIP/2.0/TCP 10.10.10.1:5093;branch=z9hG4bK-17716-1-0

Via: SIP/2.0/TCP 10.10.14.20:5093;branch=z9hG4bK-28280-1-0

 

In my situation, I’d been using a voice class uri that looked something like this:

!
voice class uri ITSP sip
    host ipv4:10.10.14.20
!
dial-peer voice tag voip
 session protocol sipv2
 incoming uri via ITSP
!
I would have expected the feature to consider the ordered list in decreasing priority, but the dial-peer never matched…  As per the feature description, only the “neighboringSIPentity” is considered.
Just something to note when deploying this feature in future.
#dontcalltac
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