The Back-to-Back User Agent (B2BUA) is a Session Initiation Protocol (SIP) call-controlling component. Unlike a SIP proxy server, which only maintains a transaction state, the B2BUA maintains the complete call state and participates in all call requests. For this reason, the B2BUA can perform a number of functions that are not possible to implement using the SIP proxy, such as accurate call accounting, pre-paid rating and billing, failover call routing, and so on.
The B2BUA is involved in call establishment, management, and termination.
B2BUA High Level Architecture
The B2BUA consists of the following three main logical components:
- Answering SIP User Agent
- Call Control Logic
- Originating SIP User Agent
The following diagram illustrates the B2BUA architecture and the primary components.
Figure 1 — B2BUA High Level Architecture
The components interact with each other using abstract events. Each User Agent (UA) represents a state machine, which receives SIP messages from the endpoint and converts them into events based on the type of message and the agent’s own current state. The Call Control Logic acts as a go-between, passing events between the UAs. Depending on its own current state and the states of the UAs, the Call Control Logic could drop some events, convert even type in transition, or inject additional events. It can also remove one of the UAs and replace it with another one on different stages of the call, which allows implementing such features as failover call routing, controlled call transfer and so on.
Since the number of parameters passed by each event is well defined the B2BUA can isolate call legs from each other, allowing only controlled amount of information to pass through.
This architecture allows implementing different functionality by replacing the Call Control Logic, which consists of a small fraction of the B2BUA code. Two such implementations are described in the next section.
Typicall Call Process
- A call is initiated when the Answering SIP UA receives an incoming INVITE message from the Originating SIP Endpoint.
- After receiving this message, the Answering SIP UA generates a Try event (2) and passes it to the Call Control Logic, as illustrated in the following diagram.
Figure 2 — Try Event (2) passed to Call Control Logic
- The Call Control Logic receives the Try event, performs authentication and authorization, creates the Originating SIP UA, modifies the Try event to accommodate for any parameter translation logic, and passes it along with the routing information to the Originating SIP UA (3).
- The Originating SIP UA receives the Try event and generates INVITE message (4) as shown in the following diagram.
Figure 3 — Originating SIP UA generating INVITE Message (4)
- After the Answering SIP endpoint receives the INVITE message, it starts ringing and sends back an 18x SIP provisional response (5).
- The Originating SIP UA receives the message, generates a Ringing event, and passes it to the Call Control Logic (6).
- The Call Control Logic receives the Ringing event and passes it to the Answering SIP UA (7), and in response, the Answering SIP UA sends an 18x SIP provisional response to the Originating SIP endpoint (8).
Figure 4 — Answering SIP UA sending response to Originating SIP endpoint
- When the user at the Answering SIP endpoint picks up the phone, the endpoint generates a 200 OK SIP response and sends it back to the Originating SIP UA (9).
- The UA generates a Connect event and passes it to the Call Control Logic (10), following which the Call Control Logic hands the event over to the Answering SIP UA (11).
- The UA sends a 200 OK message to the Originating SIP endpoint (12). At this point, the session is established and endpoints start exchanging RTP media (13).
Figure 5 — Endpoints receiving RTP Media
- When either party hangs up, the respective SIP endpoint generates a SIP BYE message and sends the message to the associated SIP UA (14).
- The UA generates a Disconnect event, which propagates to the other side of the B2BUA via the Call Control Logic (15), (16) and results in a BYE message, which is sent to the other endpoint (17).
Figure 6 — BYE message sent to Originating SIP endpoint
- The session ends.
Simple B2BUA represents very basic SIP back to back user agent. It accepts incoming SIP calls on the specified IP address and for each incoming call leg establishes outgoing call leg to the pre-defined IP address. It doesn't perform any authentication or authorization for incoming calls and passes all main parameters (CLD, CLI, SDP body, Caller Name, Callee Name) from incoming call leg to outgoing call leg verbatim. The main purpose of this B2BUA is to serve as an example for building more complex call control logic.
To invoke the B2BUA use the following command:
python b2bua_simple.py [-f] [-l local_ip] [-p local_port] [-n remote_ip]
Options enclosed in square brackets are optional. The following options are available:
- -f - run in the foreground (by default B2BUA becomes daemon after invocation);
- -l local_ip – specific IP address for listening for incoming SIP messages at;
- -p local_port – specific UDP port number for listening for incoming SIP messages at (by default the B2BUA uses port 5060);
- -n remote_ip - IP address of the target for the outgoing call legs.
For example, the following command will instruct B2BUA to listen for incoming SIP calls on IP address 192.168.0.15 and forward all outgoing calls to SIP endpoint with IP address of 192.168.1.25:
python b2bua_simple.py -l 192.168.0.15 -n 192.168.1.25
There are several environment variables that can be used to control how B2BUA logs its SIP-related activities:
- SIPLOG_BEND – specifies which method of reporting to use. Available methods are logfile and stderr (default);
- SIPLOG_LVL – specifies logging level, that is how detailed the log file should be. The following levels are available: DBUG, INFO (default), WARN, ERR, CRIT (from the most verbose to the least verbose);
- SIPLOG_LOGFILE_FILE – when logfile method is selected allows specifying location of the logfile (/var/log/sip.log by default).
RADIUS B2BUA represents advanced SIP back to back user agent designed to be used with RFC2865/RFC2866-compliant RADIUS billing engines. It accepts incoming SIP calls on the specified IP address, performs authentication and authorization using RADIUS protocol and if it has been successful establishes outgoing call leg to either pre-defined IP address or address returned by the RADIUS server in a custom attribute. It also allows RADIUS server to alter number of parameters in the outgoing call leg (CLD, CLI, Caller Name, Callee Name). Once the call has been completed or has been failed it can send RADIUS accounting.
The B2BUA uses RADIUS AAA attributes as per Cisco CDR Accounting for Cisco IOS Voice Gateways, which provides compatibility with many billing platforms. The authentication could be performed either using Cisco-compatible Remote IP method, or by using secure digest method proposed in RADIUS Extension for Digest Authentication IEFT draft.
python b2bua_radius.py [-fDu] [–l local_ip] [-p local_port] [-P pidfile] [-L logfile] [-s static_route] [-a ip1[,..[,ipN]]] [-k 0-3] [-m max_ctime] [-A 0-2] [-r rtp_proxy_contact1] [-r rtp_proxy_contact2] … [-r rtp_proxy_contactN]
Options enclosed in square brackets are optional. The following options are available:
- -f - run in the foreground (by default, B2BUA becomes the daemon after invocation)
- -D - do not use secure digest authentication
- -u - disable (do not send) RADIUS Authentication/Authorization?. If this flag is specified, the B2BUA does not send any Authentication/Authorization? requests to the RADIUS server. Unless the –a option is also specified, the B2BUA in this mode will accept incoming sessions from any IP address without any limitations. This flag depends on the –s flag, since in this case, B2BUA is not able to get routing information from the RADIUS replies.
- -A 0-2 - RADIUS accounting level. Argument in the range 0-2 specifies the level as follows:
- 0 – disable (do not send) RADIUS accounting
- 1 – enable sending Stop RADIUS accounting records (this mode is default)
- 2 – enable sending both Start and Stop RADIUS accounting records
- -l local_ip - specific IP address to listen for incoming SIP messages
- -p local_port - specific UDP port number to listen for incoming SIP messages (by default, B2BUA uses port 5060)
- -P pidfile - path to the pidfile, that is, the file that contains the PID of the B2BUA (by default, B2BUA uses /var/run/b2bua.pid)
- -L logfile - path to the file into which B2BUA should redirect its stdout and stderr (by default, B2BUA uses /var/log/b2bua.log)
- -s static_route - instead of expecting RADIUS returning routing information in reply, use static_route as the single “static” route. Note See the section Call Routing for the exact format of the static_route parameter
- -a ip1[,..[,ipN]] - accept incoming calls only from the IP addresses specified in the ip1[,..[,ipN]] list
- -k 0-3 - enable keep-alives, or re-INVITE messages, which are periodically sent to both parties participating in a call in order to detect if either party goes offline. Argument in the range 0-3 specifies mode:
- 0 – keep-alives disabled (default)
- 1 – enabled for answering call leg
- 2 – enabled for originate call leg
- 3 – enabled for both call legs
- -m max_ctime - limit maximum duration of all calls to max_ctime seconds
- -r rtp_proxy_contact - force media for all calls to go through RTP Proxy media relay specified by the rtp_proxy_contact parameter. The rtp_proxy_contact can either be the path to the local unix-domain socket at which the RTP Proxy listens for commands, or the address of the remote RTP Proxy in the format udp:hostname[:port]
- -F pt1[,...[,ptN]] - list of numeric RTP payload types ( RFC3551)that should only be allowed in the SDP offer of INVITE that the B2BUA sends to the egress call leg. Essentially this means that the B2BUA won't pass any payload types not in this list preventing answering party from using them. In the case when ingress INVITE doesn't have any payload types from this list in the SDP offer the request would be rejected with 488 Not Acceptable Here response
- -R radconf_path - path to radiusclient.conf
- -h header1[,...[,headerN]] - list of SIP header field names that the B2BUA should pass verbatim from ingress to egress call leg
- -c cmd_path - path to the control socket.
If the port is not specified, a default port value (22222) is used. It is possible to specify the –r option multiple times, in which case, B2BUA will select a particular RTP Proxy randomly for each call, effectively distributing the load evenly among all of them. In addition, the B2BUA periodically tests for and keeps track of the accessibility of each RTP Proxy and avoids sending new calls to ones that are not accessible at the moment.
The B2BUA routes incoming calls based on dynamic information returned by the RADIUS server with each authentication accept response, or alternatively, by statically using information provided in the command line. In the former case, it is expected that any positive response should contain one or more routing entries placed into the h323-ivr-in the Cisco VSA attribute. The format of the routing entries is as follows:
Options in square brackets are optional parameters. The underlined words denote variables to be filled in by the RADIUS server. If more than a single routing entry is present, routing entries will be tried one by one until either there are no more routes left or the call is successfully connected. The following parameters are available:
- cld - CLD to be used for the outgoing call
- hostname - IP address or DNS name of the SIP peer to which the call must be sent
- port - UDP port at which the SIP peer to which the call must be sent accepts SIP signaling messages
- credit-time - maximum session time to be allowed before the call is forcefully disconnected by the B2BUA
- expires - maximum time to wait until the call sent to a particular route is connected. If this time has been exceeded, the B2BUA proceeds to processing the next route if one or more routes is available, or reports a failure condition to the caller if not available.
- auth - username/password pair for SIP authentication with a remote SIP peer at hostname
- cli - CLI to be used for the outgoing call
- ash - insert arbitrary SIP header field into INVITE of the originate call leg. The parameter value should be a valid SIP header field in the format Name:Value, transformed using URL quoting rules set forth in RFC 1738.
- np_expires - maximum time to wait until non-100 provisional response on the call sent to a particular route is received or the call is connected (whichever happens first). If this time has been exceeded, the B2BUA proceeds to processing the next route if one or more routes is available, or reports a failure condition to the caller if not available.
The following is an example of a routing string in the RADIUS attribute:
h323-ivr-in = 'Routing:firstname.lastname@example.org;cli=16046288900;rid=-1;expires=30;np_expires=5;ash=Name%3AValue'
The same as the static route in the command line would be:
b2bua_radius.py ... -s 'email@example.com;cli=16046288900;rid=-1;expires=30;np_expires=5;ash=Name%3AValue' ...
Why using Python? Isn't it slow?
For certain class of application Python provides adequate performance. Being 100% text based protocol, SIP appears to be one of such applications. Also, when telecommunication applications are considered some other factors, such as reliability and resilence, have much more importance than pure performance. With the performance numbers outlined above (150-200 calls/second) one server running Sippy B2BUA can handle tenth of millions billable minites per month. Any network that operates with such vast amount of traffic has no choice but become distributed to grow further, which limits usefullness of using unsafe languages such as C or C++ to improve performance of a single B2BUA instance beyond that point.
Can the Sippy B2BUA determine if one of the peer in a session gone without a BYE message (eg. disconnected the network interface) and then send a BYE message to the other peer?
Yes, it's possible. There are two methods for determining that the one of the parties is gone: the first is by sending periodical re-INVITE to both parties (so-called SIP keep-alives), and another one is by monitoring state of the RTP session in the proxy. The first one is already supported by the B2BUA.
Installation and Configuration
How to install and configure RADIUS client?
For radiusclient-ng you should do the following:
- Install radiusclent-ng from source
~# tar xvfz radiusclient-ng-X.Y.Z.tar.gz ~# cd radiusclient-ng-X.Y.Z ~# ./configure ~# make ~# make install
- Edit /usr/local/etc/radiusclient-ng/radiusclient.conf and set address of authentication and accounting servers
authserver homero.lucio01.net acctserver homero.lucio01.net
- Edit /usr/local/etc/radiusclient-ng/servers and add shared secret for each server the client comunicates with.
- Include dictionary included in sippy b2bua dist into radiusclient-ng by copying dictionary file from Sippy distribution into /usr/local/etc/radiusclient-ng.
How to install and configure RADIUS server?
There are many RADIUS servers, both open source and commercial ones. You should refer to documentation of the selected server software on how to install and configure it. For a good GPL RADIUS server you can check FreeRADIUS.
ImportError: No module named twisted.internet when trying to run B2BUA
The B2BUA responds with 403 Auth Failed to all incoming calls, what's wrong?
There are two possible reasons for this negative response:
- RADIUS server rejects authentication request. In this case you should check the RADIUS server logs to find a problem.
- RADIUS client is not configured properly. This case can be identified by the following message in the B2BUA log:
07 Jan 19:51:firstname.lastname@example.org/b2bua: Error sending AAA request (delay is 0.003)
You should check your system log, usually /var/log/messages, to see detailed error(s) generated by the RADIUS client in this case.
- Figure 1 — B2BUA High Level Architecture.png (22.6 KB) - added by sobomax 8 years ago.
- Figure 2 — Try Event (2) passed to Call Control Logic.png (16.6 KB) - added by sobomax 8 years ago.
- Figure 3 — Originating SIP UA generating INVITE Message (4).png (21.3 KB) - added by sobomax 8 years ago.
- Figure 4 — Answering SIP UA sending response to Originating SIP endpoint.png (23.5 KB) - added by sobomax 8 years ago.
- Figure 5 — Endpoints receiving RTP Media.png (25.0 KB) - added by sobomax 8 years ago.
- Figure 6 — BYE message sent to Originating SIP endpoint.png (23.4 KB) - added by sobomax 8 years ago.