Mojo-

I'm surprised at the replies you received. Normally the people on this group are extremely helpful.

Clearly you've spent significant time thinking about this, and your current system and problem description below is detailed and well-presented, with a clear rationale for using Kamailio.

As you've already successfully installed Kamailio, my suggestion would be to continue configuring that system as a parallel (shadow) system to your existing system and start experimenting with basic call routing. Just take it one step at a time, and as you encounter specific questions, post here and I'm sure you'll get great help.

David -- I'm sorry, but as one of the old guys on here, I have to stand up and say that really was an unfair response. After a new group member writes a multi-page description clearly stating he's already using a legacy commercial SIP proxy, you don't think he's already thought about a consultant and/or commercial software? C'mon.

-Jeff

Apologies if any of the questions below are a bit dumb - I don't pretend to be an expert in SIP/VOIP - I am just an ordinary user looking for answers.

Our current setup involves processing a small number of internal sip accounts (up to 10, no more than that) and one "public" one (with a separate registrar) in the following way:

On our server we have three interfaces: eth0, eth1 and tun0. eth0 is our entry point to the public internet, eth1 faces our internal network and tun0 is a private vpn, which connects all our smartphones to the internal network (via Wifi, EDGE/2G/3G etc). This gives us the mobility.

Up until now, we have been routing voip calls via a commercial (closed source), very limited, terribly outdated (Pentium code base!) and rather buggy sip proxy. I had to employ a lot of hacks on our server in order to route calls as this proxy can only listen on a single interface. It was also a nightmare to maintain. Unsurprisingly, I decided that enough is enough and I am now determined to replace it.

We route calls in the following way: all machines (PCs are all Linux based) & smartphones have their own sip/voip client installed on them (also using bluetooth). Internal calls are routed via the proxy between ourselves either on the internal net (eth1), or between the vpn and eth1 (eth1<->tun0).

External calls (going out, i.e. outbound) are routed externally to our registrar, using a single separate voip account, via eth1<->eth0 or tun0<->eth0.

As I am now looking to replace our proxy, I looked at Kamailio, but was soon completely overwhelmed by it (no offence intended, it was just too much to take at first). I would appreciate if any of you could give me a hand, or at least point me in the right direction, with the following issues:

I presume I could configure Kamailio to listen on more than one interface and act as a proxy. How do I do that, so that it listens on all 3 interfaces and proxies requests in the following way:

• calls made to <userX>@ourdomain.net to be routed internally via eth1

(internal net) or tun0 (private vpn);

• calls made to anybody else to be routed externally via eth0 (public)

using the separate "public" sip account with our external registrar;

• calls made to the public sip account (from outside - the "public")

need to be routed to a "nominated" internall account (say <user0>@ourdomain.net);

• all other (internal) calls need to be routed depending on which

interface this account has been registered/logged in - either the internal net (eth1) or the private vpn (tun0 - the smartphones).

Obviously, calls need to be received (and routed properly) from all 3 interfaces.

Is all of this possible with Kamailio?

I want to avoid unnecessary complexities of the setup (as I already mentioned above - I am just a user and by no means an expert in sip/voip) and do not want to deploy something I do not need - I need to keep the memory footprint to a bare minimum, possibly without sacrificing performance.

Once this is done, I would then move on to the next phase and use IM & ENUM, but this is once the above works.

I looked at other alternatives, but I got very confused there as well - I couldn't figure out what exactly is the difference between, say, OpenSER, Kamailio, OpenSIPS and SIP-Router even? What is the best software to use in order to achieve the above setup?

One last thing - I am a developer by trade and I am not afraid of "tweaking" things when needed. I was successful in compiling Kamailio from source (I use Fedora on all our machines) and I was pleased that I could exclude from the RPM .spec file the modules I think I did not need.

I also made some modification of my own to make the database modules (mysql, postgresql and unixodbc) configurable in the same way the rest of the modules are. I could submit patches, if needed, so that these are incorporated into future releases - how do I do that?

I could not do the same with OpenSIPS, however (which I also tried - out of curiosity!) - everything there seems to be lumped and compiled together regardless of whether it is needed or not.

Any help as to helping me with the above issues is greatly appreciated, many thanks in advance for taking the time!

---

SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

All right, fair enough. I realise that this list has left you with little useful takeaways thus far, and, upon further reflection, that does seem a little unfair, given the amount of effort and thought that went into your post.

In due recognition of that, let me _try_ to tackle some of the more "low-hanging fruit" here, bearing in mind everything I've had to say about the practicality of the scope of your questions.

See below:

On 01/30/2012 12:00 PM, Me wrote:

External calls (going out, i.e. outbound) are routed externally to our registrar, using a single separate voip account, via eth1<->eth0 or tun0<->eth0.

As I am now looking to replace our proxy, I looked at Kamailio, but was soon completely overwhelmed by it (no offence intended, it was just too much to take at first). I would appreciate if any of you could give me a hand, or at least point me in the right direction, with the following issues:

It is understandable that you were overwhelmed. It is a relatively low-level, idiosyncratic software product, with its own programmatic route script language for configuration, and many associated, idiosyncratic APIs and conventions. The overall flow of the route script mimics aspects of the SIP state machine, and is tied to SIP message processing. You can think of Kamailio as something like an SDK on top of a proxy core. So, for starters, a fairly robust understanding of SIP protocol mechanics is required.

As much as efforts have been made to make it more user-friendly, it is something that really has little out-of-the-box functionality suitable for all but the most generic of implementation scenarios.

Roughly speaking, the stock configuration file is, to some extent by convention and to some extent by necessity, organised into three parts:

1. Global configuration values. 2. Module loading & module-specific parameters. 3. Routes.

Routes are like functions or subroutines. They are activated when a SIP message is processed. The canonical route is a request route, so it is fired when a SIP request is received (initial or sequential/in-dialog). There are other types of routes for SIP message types other than requests, like replies, and also for certain conditions that inhere in Kamailio's design, such as failure routes (for >= 3xx failure replies), event routes (functionality-specific callbacks), etc.

At the bare minimum, to do any routing, you need the single mandatory request route, which has no name:

route { ... }

All SIP requests pass through this main request route.

You can delegate functionality into other routes which can be called from the main request route, e.g.

route { ...

route(X); }

route[X] { ... }

All other types of routes are pretty much optional, and not essential to a "quick start" or getting going.

The root of all documentation is here:

http://www.kamailio.org/dokuwiki/doku.php

There are several components to the documentation tree:

- Core cookbook: core functionality, including route types, global configuration values, variables, primitives, operators, etc.

- Modules: http://www.kamailio.org/docs/modules/3.2.x/

Modules are optionally loadable and provide all non-core functionality. Note that some components of Kamailio are technically modular, but in practice, are essential to almost any commonplace use of it, so they can be regarded as core. These include: rr, xlog, tm, pv, sl. Your best guide for this aspect of things is the stock config file.

- Pseudovariables & selects: these are variables which are populated by Kamailio automatically to expose SIP message attributes. Most of these are immutable (read-only), while a few, such as the request URI ($ru), are mutable, so you can actually assign values to them.

- Transformations: these are mostly string-related operations that can be performed on variables of all kinds (user variables and pseudovariables), e.g.

$(ru{s.len}) -- get the length of the request URI string.

...

Your best bet is to take the stock configuration file and tweak it to your liking. I encourage you to conserve the sanity checking and request handling boilerplate you see at the top of the main request route, as that is stuff you are going to want in virtually any imaginable scenario. In particular, don't make the common rookie mistake of deleting the

if(has_totag()) { ...

if(loose_route()) { ... } else { ... } }

section. Sequential (in-dialog) requests with a preset route set (i.e. Route header) must be routed differently than initial requests; routing then manually is poor form.

I presume I could configure Kamailio to listen on more than one interface and act as a proxy. How do I do that, so that it listens on all 3 interfaces and proxies requests in the following way:

You just bind it to all three interfaces with multiple 'listen' parameters, e.g.

listen=udp:xxx.xxx.xxx.xxx:5060 listen=udp:yyy.yyy.yyy.yyy:5060 listen=udp:zzz.zzz.zzz.zzz:5060

If you want to have Kamailio take its best shot at guessing which interface to route out of, set

mhomed=1.

If you want to guide this process manually, just modify the $fs pseudovariable prior to relaying a message, or use force_send_socket(), e.g.

$fs = "udp:xxx.xxx.xxx.xxx:5060";

# This will go out of the xxx.xxx.xxx.xxx:5060 socket.

if(!t_relay()) { sl_reply_error(); exit; }

• calls made to <userX>@ourdomain.net to be routed internally via eth1

(internal net) or tun0 (private vpn);

• calls made to anybody else to be routed externally via eth0 (public)

using the separate "public" sip account with our external registrar;

• calls made to the public sip account (from outside - the "public")

need to be routed to a "nominated" internall account (say <user0>@ourdomain.net);

• all other (internal) calls need to be routed depending on which

interface this account has been registered/logged in - either the internal net (eth1) or the private vpn (tun0 - the smartphones).

You can get the destination domain of a request by accessing the request URI attributes, e.g.

$ru - the whole request URI. $rU - the user part of the request URI. $rd - the domain part of the request URI. $rp - the port part of the request URI.

e.g.

if($rd == "ourdomain.net") route(INTERNAL_ROUTE);

Getting Kamailio to act as a UAC to register with an external registrar is a little complex, since Kamailio is a proxy, and proxies are not UACs, so they are not capable, at least, as a formal matter, of initiating requests or registering to anything. However, Kamailio has this functionality hacked in, in the form of its 'uac' module:

http://www.kamailio.org/docs/modules/3.2.x/modules_k/uac.html

In particular, there is a 'uacreg' table you can use to get it to initiate and maintain registrations with an external SIP provider. But this is a rather involved topic.

Obviously, calls need to be received (and routed properly) from all 3 interfaces.

Is all of this possible with Kamailio?

In principle, yes.

There are complications you will encounter along the way, particularly if you want Kamailio to stay in the signaling path of sequential (in-dialog) requests, which you would do using by inserting the Record-Route header using the 'rr' module's record_route() function. Multiple interfaces require careful handling.

If you don't care about the proxy seeing BYEs or reinvites, your life will be much simpler in a multihomed scenario.

I looked at other alternatives, but I got very confused there as well

• I couldn't figure out what exactly is the difference between, say,

OpenSER, Kamailio, OpenSIPS and SIP-Router even? What is the best software to use in order to achieve the above setup?

This is a complex and somewhat politically charged topic, but in general, the lineage is something like this:

OpenSER | | | | Kamailio | \ | \ | \ | OpenSIPS | Kamailio+sip-router

SER is the common ancestor to this entire technology stack. OpenSER is the open-source variant that appeared in 2005. In 2008, its name was changed to Kamailio due to trademark-related issues, and at about the same time, OpenSIPS was forked from it. OpenSIPS is considered a rogue fork around here, and by this point has diverged significantly. I don't know that this list is the best place to get an opinion on the possible benefits of using OpenSIPS, but I can tell you that most of the people involved in managing the hitherto joint OpenSER project stayed with the Kamailio effort and, in my opinion, it has a lot more industry power and ecosystem behind it.

The sip-router project is an initiative that resulted in the recombination of Kamailio with the original SER, to leverage the best of both worlds. You can think of sip-router as a "kernel" and Kamailio as a particular packaging/distribution of it. For all practical purposes, they are the same thing, as far as you should be concerned. In other words, Kamailio has sip-router underneath.

-- Alex