2. Architecture

2.1. Platforms

2.1.1. CE Platform

The Sipwise C5 CE platform is one single node running all necessary components of the system. The components are outlined in the following figure:

CE Architecture Overview

Figure 1. CE Architecture Overview

The main building blocks of Sipwise C5 are:

  • Provisioning
  • SIP Signaling and Media Relay
  • Mediation and Billing

2.2. SIP Signaling and Media Relay

In SIP-based communication networks, it is important to understand that the signaling path (e.g. for call setup and tear-down) is completely independent of the media path. On the signaling path, the involved endpoints negotiate the call routing (which user calls which endpoint, and via which path - e.g. using SIP peerings or going through the PSTN - the call is established) as well as the media attributes (via which IPs/ports are media streams sent and which capabilities do these streams have - e.g. video using H.261 or Fax using T.38 or plain voice using G.711). Once the negotiation on signaling level is done, the endpoints start to send their media streams via the negotiated paths.

The components involved in SIP and Media on the Sipwise C5 CE are shown in the following figure:

SIP and Media Relay Components

Figure 2. SIP and Media Relay Components

2.2.1. SIP Load-Balancer

The SIP load-balancer is a Kamailio instance acting as ingress and egress point for all SIP traffic to and from the system. It’s a high-performance SIP proxy instance based on Kamailio and is responsible for sanity checks of inbound SIP traffic. It filters broken SIP messages, rejects loops and relay attempts and detects denial-of-service and brute-force attacks and gracefully handles them to protect the underlying SIP elements. It also performs the conversion of TLS to internal UDP and vice versa for secure signaling between endpoints and Sipwise C5, and does far-end NAT traversal in order to enable signaling through NAT devices.

The load-balancer is the only SIP element in the system which exposes a SIP interface to the public network. Its second leg binds in the switch-internal network to pass traffic from the public internet to the corresponding internal components.

The name load-balancer comes from the fact that in the commercial version, when scaling out the system beyond just one pair of servers, the load-balancer instance becomes its own physical node and then handles multiple pairs of proxies behind it.

On the public interface, the load-balancer listens on port 5060 for UDP and TCP, as well as on 5061 for TLS connections. On the internal interface, it speaks SIP via UDP on port 5060 to the other system components, and listens for XMLRPC connections on TCP port 5060, which can be used to control the daemon.

Its config files reside in /etc/ngcp-config/templates/etc/kamailio/lb/, and changes to these files are applied by executing ngcpcfg apply "my commit message".


The SIP load-balancer can be managed via the commands ngcp-service start kamailio-lb, ngcp-service stop kamailio-lb and ngcp-service restart kamailio-lb. Its status can be queried by executing ngcp-service status kamailio-lb or ngcp-service summary | grep "kamailio-lb". Also ngcp-kamctl lb and ngcp-kamcmd lb are provided for querying kamailio functions, for example: ngcp-kamcmd lb htable.dump ipban. Execute the command: ngcp-kamctl lb fifo system.listMethods or ngcp-kamcmd lb system.listMethods to get the list of all available queries.

2.2.2. SIP Proxy/Registrar

The SIP proxy/registrar (or short proxy) is the work-horse of Sipwise C5. It’s also a separate Kamailio instance running in the switch-internal network and is connected to the provisioning database via MySQL, authenticates the endpoints, handles their registrations on the system and does the call routing based on the provisioning data. For each call, the proxy looks up the provisioned features of both the calling and the called party (either subscriber or domain features if it’s a local caller and/or callee, or peering features if it’s from/to an external endpoint) and acts accordingly, e.g. by checking if the call is blocked, by placing call-forwards if applicable and by normalizing numbers into the appropriate format, depending on the source and destination of a call.

It also writes start- and stop-records for each call, which are then transformed into call detail records (CDR) by the mediation system.

If the endpoints indicate negotiation of one or more media streams, the proxy also interacts with the Media Relay to open, change and close port pairs for relaying media streams over Sipwise C5, which is especially important to traverse NAT.

The proxy listens on UDP port 5062 in the system-internal network. It cannot be reached directly from the outside, but only via the SIP load-balancer.

Its config files reside in /etc/ngcp-config/templates/etc/kamailio/proxy/, and changes to these files are applied by executing ngcpcfg apply "my commit message".


The SIP proxy can be controlled via the commands ngcp-service start kamailio-proxy, ngcp-service stop kamailio-proxy and ngcp-service restart kamailio-proxy. Its status can be queried by executing ngcp-service status kamailio-proxy or ngcp-service summary | grep "kamailio-proxy". Also ngcp-kamctl proxy and ngcp-kamcmd proxy are provided for querying kamailio functions, for example: ngcp-kamctl proxy ul show. Execute the command: ngcp-kamctl proxy fifo system.listMethods or ngcp-kamcmd proxy system.listMethods to get the list of all available queries.

2.2.3. SIP Back-to-Back User-Agent (B2BUA)

The SIP B2BUA (also called SBC within the system) decouples the first call-leg (calling party to Sipwise C5) from the second call-leg (Sipwise C5 to the called party).

The software part used for this element is SEMS.

This element is typically optional in SIP systems, but it is always used for SIP calls (INVITE) that don’t have Sipwise C5 as endpoint. It acts as application server for various scenarios (e.g. for feature provisioning via Vertical Service Codes and as Conferencing Server) and performs the B2BUA decoupling, topology hiding, caller information hiding, SIP header and Media feature filtering, outbound registration, outbound authentication and call length limitation as well as Session Keep-Alive handler.

Due to the fact that typical SIP proxies (like the load-balancer and proxy in Sipwise C5) do only interfere with the content of SIP messages where it’s necessary for the SIP routing, but otherwise leave the message intact as received from the endpoints, whereas the B2BUA creates a new call leg with a new SIP message from scratch towards the called party, SIP message sizes are reduced significantly by the B2BUA. This helps to bring the message size under 1500 bytes (which is a typical default value for the MTU size) when it leaves Sipwise C5. That way, chances of packet fragmentation are quite low, which reduces the risk of running into issues with low-cost SOHO routers at customer sides, which typically have problems with UDP packet fragmentation.

The SIP B2BUA only binds to the system-internal network and listens on UDP port 5080 for SIP messages from the load-balancer or the proxy, on UDP port 5040 for control messages from the cli tool and on TCP port 8090 for XMLRPC connections to control the daemon.

Its configuration files reside in /etc/ngcp-config/templates/etc/ngcp-sems, and changes to these files are applied by executing ngcpcfg apply "my commit message".


The SIP B2BUA can be controlled via the commands ngcp-service start sems, ngcp-service stop sems and ngcp-service restart sems. Its status can be queried by executing ngcp-service status sems or ngcp-service summary | grep "sems".

2.2.4. SIP App-Server

The SIP App-Server is an Asterisk instance used for voice applications like Voicemail and Reminder Calls. Asterisk uses the MySQL database as a message spool for voicemail, so it doesn’t directly access the file system for user data. The voicemail plugin is a slightly patched version based on Asterisk 1.4 to make Asterisk aware of Sipwise C5 internal UUIDs for each subscriber. That way a SIP subscriber can have multiple E164 phone numbers, but all of them terminate in the same voicebox.

The App-Server listens on the internal interface on UDP port 5070 for SIP messages and by default uses media ports in the range from UDP port 10000 to 20000.

The configuration files reside in /etc/ngcp-config/templates/etc/asterisk, and changes to these files are applied by executing ngcpcfg apply "my commit message".


The SIP App-Server can be controlled via the commands ngcp-service start asterisk, ngcp-service stop asterisk and ngcp-service restart asterisk. Its status can be queried by executing ngcp-service status asterisk or ngcp-service summary | grep "asterisk".

2.2.5. Message Routing and Media Relay

The Media Relay (also called rtpengine) is a Kernel-based packet relay, which is controlled by the SIP proxy. For each media stream (e.g. a voice and/or video stream), it maintains a pair of ports in the range of port number 30000 to 40000. When the media streams are negotiated, rtpengine opens the ports in user-space and starts relaying the packets to the addresses announced by the endpoints. If packets arrive from different source addresses than announced in the SDP body of the SIP message (e.g. in case of NAT), the source address is implicitly changed to the address the packets are received from. Once the call is established and the rtpengine has received media packets from both endpoints for this call, the media stream is pushed into the kernel and is then handled by a custom Sipwise iptables module to increase the throughput of the system and to reduce the latency of media packets.

The rtpengine internally listens on UDP port 12222 for control messages from the SIP proxy. For each media stream, it opens two pairs of UDP ports on the public interface in the range of 30000 and 40000 per default, one pair on even port numbers for the media data, and one pair on the next odd port numbers for metadata, e.g. RTCP in case of RTP streams. Each endpoint communicates with one dedicated port per media stream (opposed to some implementations which use one pair for both endpoints) to avoid issues in determining where to send a packet to. The rtpengine also sets the QoS/ToS/DSCP field of each IP packet it sends to a configured value, 184 (0xB8, expedited forwarding) by default.

The kernel-internal part of the rtpengine is facilitated through an iptables module having the target name RTPENGINE. If any additional firewall or packet filtering rules are installed, it is imperative that this rule remains untouched and stays in place. Otherwise, if the rule is removed from iptables, the kernel will not be able to forward the media packets and forwarding will fall back to the user-space daemon. The packets will still be forwarded normally, but performance will be much worse under those circumstances, which will be especially noticeable when a lot of media streams are active concurrently. See the section on Firewalling for more information.

The rtpengine configuration file is /etc/ngcp-config/templates/etc/default/ngcp-rtpengine-daemon, and changes to this file are applied by executing ngcpcfg apply "my commit message". The UDP port range can be configured via the config.yml file under the section rtpproxy. The QoS/ToS value can be changed via the key qos.tos_rtp.


The Media Relay can be controlled via the commands ngcp-service start rtpengine, ngcp-service stop rtpengine and ngcp-serivce restart rtpengine. Its status can be queried by executing ngcp-service status rtpengine" or ngcp-service summary | grep "rtpengine".

2.3. Redis Database

The redis database is used as a high-perfomance key/value storage for global system data. This includes calls information and concurrent calls counters for customers and subscribers, etc..