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What is rtpengine?

The Sipwise NGCP rtpengine is a proxy for RTP traffic and other UDP based media traffic. It's meant to be used with the Kamailio SIP proxy and forms a drop-in replacement for any of the other available RTP and media proxies.

Currently the only supported platform is GNU/Linux.

Features

  • Media traffic running over either IPv4 or IPv6
  • Bridging between IPv4 and IPv6 user agents
  • Bridging between different IP networks or interfaces
  • TOS/QoS field setting
  • Customizable port range
  • Multi-threaded
  • Advertising different addresses for operation behind NAT
  • In-kernel packet forwarding for low-latency and low-CPU performance
  • Automatic fallback to normal userspace operation if kernel module is unavailable
  • Support for Kamailio's rtpproxy module
  • Legacy support for old OpenSER mediaproxy module

When used through the rtpengine module (or its older counterpart called rtpproxy-ng), the following additional features are available:

  • Full SDP parsing and rewriting
  • Supports non-standard RTCP ports (RFC 3605)
  • ICE (RFC 5245) support:
    • Bridging between ICE-enabled and ICE-unaware user agents
    • Optionally acting only as additional ICE relay/candidate
    • Optionally forcing relay of media streams by removing other ICE candidates
  • SRTP (RFC 3711) support:
    • Support for SDES (RFC 4568) and DTLS-SRTP (RFC 5764)
    • AES-CM and AES-F8 ciphers, both in userspace and in kernel
    • HMAC-SHA1 packet authentication
    • Bridging between RTP and SRTP user agents
  • Support for RTCP profile with feedback extensions (RTP/AVPF, RFC 4585 and 5124)
  • Arbitrary bridging between any of the supported RTP profiles (RTP/AVP, RTP/AVPF, RTP/SAVP, RTP/SAVPF)
  • RTP/RTCP multiplexing (RFC 5761) and demultiplexing
  • Breaking of BUNDLE'd media streams (draft-ietf-mmusic-sdp-bundle-negotiation)
  • Recording of media streams, decrypted if possible

Rtpengine does not (yet) support:

  • Repacketization or transcoding
  • Playback of pre-recorded streams/announcements
  • ZRTP, although ZRTP passes through rtpengine just fine

Compiling and Installing

On a Debian System

On a Debian system, everything can be built and packaged into Debian packages by executing dpkg-buildpackage (which can be found in the dpkg-dev package) in the main directory. This script will issue an error and stop if any of the dependency packages are not installed.

Before that, run ./debian/flavors/no_ngcp in order to remove any NGCP dependencies.

This will produce a number of .deb files, which can then be installed using the dpkg -i command.

The generated files are (with version 2.3.0 being built on an amd64 system):

  • ngcp-rtpengine_2.3.0_all.deb

    This is a meta-package, which doesn't contain or install anything on its own, but rather only depends on the other packages to be installed. Not strictly necessary to be installed.

  • ngcp-rtpengine-daemon_2.3.0_amd64.deb

    This installed the userspace daemon, which is the main workhorse of rtpengine. This is the minimum requirement for anything to work.

  • ngcp-rtpengine-dbg_2.3.0_amd64.deb

    Debugging symbols for the daemon. Optional.

  • ngcp-rtpengine-iptables_2.3.0_amd64.deb

    Installs the plugin for iptables and ip6tables. Necessary for in-kernel operation.

  • ngcp-rtpengine-kernel-dkms_2.3.0_all.deb

    Kernel module, DKMS version of the package. Recommended for in-kernel operation. The kernel module will be compiled against the currently running kernel using DKMS.

  • ngcp-rtpengine-kernel-source_2.3.0_all.deb

    If DKMS is unavailable or not desired, then this package will install the sources for the kernel module for manual compilation. Required for in-kernel operation, but only if the DKMS package can't be used.

Manual Compilation

There's 3 parts to rtpengine, which can be found in the respective subdirectories.

  • daemon

    The userspace daemon and workhorse, minimum requirement for anything to work. Running make will compile the binary, which will be called rtpengine. The following software packages including their development headers are required to compile the daemon:

    • pkg-config
    • GLib including GThread version 2.x
    • zlib
    • OpenSSL
    • PCRE library
    • XMLRPC-C version 1.16.08 or higher
    • hiredis library

    The Makefile contains a few Debian-specific flags, which may have to removed for compilation to be successful. This will not affect operation in any way.

  • iptables-extension

    Required for in-kernel packet forwarding.

    With the iptables development headers installed, issuing make will compile the plugin for iptables and ip6tables. The file will be called libxt_RTPENGINE.so and should be copied into the directory /lib/xtables/.

  • kernel-module

    Required for in-kernel packet forwarding.

    Compilation of the kernel module requires the kernel development headers to be installed in /lib/modules/$VERSION/build/, where $VERSION is the output of the command uname -r. For example, if the command uname -r produces the output 3.9-1-amd64, then the kernel headers must be present in /lib/modules/3.9-1-amd64/build/. The last component of this path (build) is usually a symlink somewhere into /usr/src/, which is fine.

    Successful compilation of the module will produce the file xt_RTPENGINE.ko. The module can be inserted into the running kernel manually through insmod xt_RTPENGINE.ko (which will result in an error if depending modules aren't loaded, for example the x_tables module), but it's recommended to copy the module into /lib/modules/$VERSION/updates/, followed by running depmod -a. After this, the module can be loaded by issuing modprobe xt_RTPENGINE.

Usage

Userspace Daemon

The daemon supports a number of command-line options, which it will print if started with the --help option and which are reproduced below:

  -v, --version                    Print build time and exit
  -t, --table=INT                  Kernel table to use
  -F, --no-fallback                Only start when kernel module is available
  -i, --interface=[NAME/]IP[!IP]   Local interface for RTP
  -l, --listen-tcp=[IP:]PORT       TCP port to listen on
  -u, --listen-udp=[IP46:]PORT     UDP port to listen on
  -n, --listen-ng=[IP46:]PORT      UDP port to listen on, NG protocol
  -c, --listen-cli=[IP46:]PORT     TCP port to listen on, CLI (command line interface)
  -g, --graphite=IP46:PORT         TCP address of graphite statistics server
  -G, --graphite-interval=INT      Graphite data statistics send interval
  --graphite-prefix=STRING         Graphite prefix for every line
  -T, --tos=INT                    TOS value to set on streams
  -o, --timeout=SECS               RTP timeout
  -s, --silent-timeout=SECS        RTP timeout for muted
  -a, --final-timeout=SECS         Call timeout
  -p, --pidfile=FILE               Write PID to file
  -f, --foreground                 Don't fork to background
  -m, --port-min=INT               Lowest port to use for RTP
  -M, --port-max=INT               Highest port to use for RTP
  -r, --redis=[PW@]IP:PORT/INT     Connect to Redis database
  -w, --redis-write=[PW@]IP:PORT/INT Connect to Redis write database
  -k, --subscribe-keyspace         Subscription keyspace list
  --redis-num-threads=INT          Number of Redis restore threads
  --redis-expires=INT              Expire time in seconds for redis keys	 
  -q, --no-redis-required          Start even if can't connect to redis databases
  -b, --b2b-url=STRING             XMLRPC URL of B2B UA
  -L, --log-level=INT              Mask log priorities above this level
  --log-facility=daemon|local0|... Syslog facility to use for logging
  --log-facility-cdr=local0|...    Syslog facility to use for logging CDRs
  --log-facility-rtcp=local0|...   Syslog facility to use for logging RTCP data (take care of traffic amount)
  -E, --log-stderr                 Log on stderr instead of syslog
  -x, --xmlrpc-format=INT          XMLRPC timeout request format to use. 0: SEMS DI, 1: call-id only
  --num-threads=INT                Number of worker threads to create
  -d, --delete-delay               Delay for deleting a session from memory.
  --sip-source                     Use SIP source address by default
  --dtls-passive                   Always prefer DTLS passive role
  --max-sessions=INT               Limit the number of maximum concurrent sessions
  --homer=IP46:PORT                Address of Homer server for RTCP stats
  --homer-protocol=udp|tcp         Transport protocol for Homer (default udp)
  --homer-id=INT                   'Capture ID' to use within the HEP protocol
  --recording-dir=FILE             Spool directory where PCAP call recording data goes
  --recording-method=pcap|proc     Strategy for call recording
  --recording-format=raw|eth       PCAP file format for recorded calls.

Most of these options are indeed optional, with two exceptions. It's mandatory to specify at least one local IP address through --interface, and at least one of the --listen-... options must be given.

The options are described in more detail below.

  • -v, --version

    If called with this option, the rtpengine daemon will simply print its version number and exit.

  • -t, --table

    Takes an integer argument and specifies which kernel table to use for in-kernel packet forwarding. See the section on in-kernel operation for more detail. Optional and defaults to zero. If in-kernel operation is not desired, a negative number can be specified.

  • -F, --no-fallback

    Will prevent fallback to userspace-only operation if the kernel module is unavailable. In this case, startup of the daemon will fail with an error if this option is given.

  • -i, --interface

    Specifies a local network interface for RTP. At least one must be given, but multiple can be specified. The format of the value is [NAME/]IP[!IP] with IP being either an IPv4 address or an IPv6 address.

    The second IP address after the exclamation point is optional and can be used if the address to advertise in outgoing SDP bodies should be different from the actual local address. This can be useful in certain cases, such as your SIP proxy being behind NAT. For example, --interface=10.65.76.2!192.0.2.4 means that 10.65.76.2 is the actual local address on the server, but outgoing SDP bodies should advertise 192.0.2.4 as the address that endpoints should talk to. Note that you may have to escape the exlamation point from your shell, e.g. using \!.

    Giving an interface a name (separated from the address by a slash) is optional; if omitted, the name default is used. Names are useful to create logical interfaces which consist of one or more local addresses. It is then possible to instruct rtpengine to use particular interfaces when processing an SDP message, to use different local addresses when talking to different endpoints. The most common use case for this is to bridge between one or more private IP networks and the public internet.

    For example, if clients coming from a private IP network must communicate their RTP with the local address 10.35.2.75, while clients coming from the public internet must communicate with your other local address 192.0.2.67, you could create one logical interface pub and a second one priv by using --interface=pub/192.0.2.67 --interface=priv/10.35.2.75. You can then use the direction option to tell rtpengine which local address to use for which endpoints (either pub or priv).

    If multiple logical interfaces are configured, but the direction option isn't given in a particular call, then the first interface given on the command line will be used.

    It is possible to specify multiple addresses for the same logical interface (the same name). Most commonly this would be one IPv4 addrsess and one IPv6 address, for example: --interface=192.168.63.1 --interface=fe80::800:27ff:fe00:0. In this example, no interface name is given, therefore both addresses will be added to a logical interface named default. You would use the address family option to tell rtpengine which address to use in a particular case.

    It is also possible to have multiple addresses of the same family in a logical network interface. In this case, the first address (of a particular family) given for an interface will be the primary address used by rtpengine for most purposes. Any additional addresses will be advertised as additional ICE candidates with increasingly lower priority. This is useful on multi-homed systems and allows endpoints to choose the best possible path to reach the RTP proxy. If ICE is not being used, then additional addresses will go unused.

    If you're not using the NG protocol but rather the legacy UDP protocol used by the rtpproxy module, the interfaces must be named internal and external corresponding to the i and e flags if you wish to use network bridging in this mode.

  • -l, --listen-tcp, -u, --listen-udp, -n, --listen-ng

    These options each enable one of the 3 available control protocols if given and each take either just a port number as argument, or an address:port pair, separated by colon. At least one of these 3 options must be given.

    The tcp protocol is obsolete. It was used by old versions of OpenSER and its mediaproxy module. It's provided for backwards compatibility.

    The udp protocol is used by Kamailio's rtpproxy module. In this mode, rtpengine can be used as a drop-in replacement for any other compatible RTP proxy.

    The ng protocol is an advanced control protocol and can be used with Kamailio's rtpengine module. With this protocol, the complete SDP body is passed to rtpengine, rewritten and passed back to Kamailio. Several additional features are available with this protocol, such as ICE handling, SRTP bridging, etc.

    It is recommended to specify not only a local port number, but also 127.0.0.1 as interface to bind to.

  • -c, --listen-cli

    TCP ip and port to listen for the CLI (command line interface).

  • -g, --graphite

    Address of the graphite statistics server.

  • -w, --graphite-interval

    Interval of the time when information is sent to the graphite server.

  • --graphite-prefix

    Add a prefix for every graphite line.

  • -t, --tos

    Takes an integer as argument and if given, specifies the TOS value that should be set in outgoing packets. The default is to leave the TOS field untouched. A typical value is 184 (Expedited Forwarding).

  • -o, --timeout

    Takes the number of seconds as argument after which a media stream should be considered dead if no media traffic has been received. If all media streams belonging to a particular call go dead, then the call is removed from rtpengine's internal state table. Defaults to 60 seconds.

  • -s, --silent-timeout

    Ditto as the --timeout option, but applies to muted or inactive media streams. Defaults to 3600 (one hour).

  • -a, --final-timeout

    The number of seconds since call creation, after call is deleted. Useful for limiting the lifetime of a call. This feature can be disabled by setting the parameter to 0. By default this timeout is disabled.

  • -p, --pidfile

    Specifies a path and file name to write the daemon's PID number to.

  • -f, --foreground

    If given, prevents the daemon from daemonizing, meaning it will stay in the foreground. Useful for debugging.

  • -m, --port-min, -M, --port-max

    Both take an integer as argument and together define the local port range from which rtpengine will allocate UDP ports for media traffic relay. Default to 30000 and 40000 respectively.

  • -L, --log-level

    Takes an integer as argument and controls the highest log level which will be sent to syslog. The log levels correspond to the ones found in the syslog(3) man page. The default value is 6, equivalent to LOG_INFO. The highest possible value is 7 (LOG_DEBUG) which will log everything.

    During runtime, the log level can be decreased by sending the signal SIGURS1 to the daemon and can be increased with the signal SIGUSR2.

  • --log-facilty=daemon|local0|...|local7|...

    The syslog facilty to use when sending log messages to the syslog daemon. Defaults to daemon.

  • --log-facilty-cdr=daemon|local0|...|local7|...

    Same as --log-facility with the difference that only CDRs are written to this log facility.

  • --log-facilty-rtcp=daemon|local0|...|local7|...

    Same as --log-facility with the difference that only RTCP data is written to this log facility. Be careful with this parameter since there may be a lot of information written to it.

  • -E, --log-stderr

    Log to stderr instead of syslog. Only useful in combination with --foreground.

  • --num-threads

    How many worker threads to create, must be at least one. The default is to create as many threads as there are CPU cores available. If the number of CPU cores cannot be determined, the default is four.

  • --sip-source

    The original rtpproxy as well as older version of rtpengine by default didn't honour IP addresses given in the SDP body, and instead used the source address of the received SIP message as default endpoint address. Newer versions of rtpengine reverse this behaviour and honour the addresses given in the SDP body by default. This option restores the old behaviour.

  • --dtls-passive

    Enables the DTLS=passive flag for all calls unconditionally.

  • -d, --delete-delay

    Delete the call from memory after the specified delay from memory. Can be set to zero for immediate call deletion.

  • -r, --redis

    Connect to specified Redis database (with the given database number) and use it for persistence storage. The format of this option is ADDRESS:PORT/DBNUM, for example 127.0.0.1:6379/12 to connect to the Redis DB number 12 running on localhost on the default Redis port.

    If the Redis database is protected with an authentication password, the password can be supplied by prefixing the argument value with the password, separated by an @ symbol, for example [email protected]:6379/12. Note that this leaves the password visible in the process list, posing a security risk if untrusted users access the same system. As an alternative, the password can also be supplied in the shell environment through the environment variable RTPENGINE_REDIS_AUTH_PW.

    On startup, rtpengine will read the contents of this database and restore all calls stored therein. During runtime operation, rtpengine will continually update the database's contents to keep it current, so that in case of a service disruption, the last state can be restored upon a restart.

    When this option is given, rtpengine will delay startup until the Redis database adopts the master role (but see below).

  • -w, --redis-write

    Configures a second Redis database for write operations. If this option is given in addition to the first one, then the first database will be used for read operations (i.e. to restore calls from) while the second one will be used for write operations (to update states in the database).

    For password protected Redis servers, the environment variable for the password is RTPENGINE_REDIS_WRITE_AUTH_PW.

    When both options are given, rtpengine will start and use the Redis database regardless of the database's role (master or slave).

  • -k, --subscribe-keyspace

    List of redis keyspaces to subscribe. If this is not present, no keyspaces are subscribed (default behaviour). Further subscriptions could be added/removed via 'rtpengine-ctl ksadd/ksrm'. This may lead to enabling/disabling of the redis keyspace notification feature.

  • --redis-num-threads

    How many redis restore threads to create. The default is four.

  • --redis-expires

     Expire time in seconds for redis keys. Default is 86400.

  • -q, --no-redis-required When this paramter is present or NO_REDIS_REQUIRED='yes' or '1' in config file, rtpengine starts even if there is no initial connection to redis databases(either to -r or to -w or to both redis).

    Be aware that if the -r redis can't be initially connected, sessions are not reloaded upon rtpengine startup, even though rtpengine still starts.

  • -b, --b2b-url

    Enables and sets the URI for an XMLRPC callback to be made when a call is torn down due to packet timeout. The special code %% can be used in place of an IP address, in which case the source address of the originating request will be used.

  • -x, --xmlrpc-format

    Selects the internal format of the XMLRPC callback message for B2BUA call teardown. 0 is for SEMS, 1 is for a generic format containing the call-ID only.

  • --max-sessions

    Limit the number of maximum concurrent sessions. Set at startup via MAX_SESSIONS in config file. Set at runtime via rtpengine-ctl util. Setting the 'rtpengine-ctl set maxsessions 0' can be used in draining rtpengine sessions. Enable feature: 'MAX_SESSIONS=1000' Enable feature: 'rtpengine-ctl set maxsessions' >=0 Disable feature: 'rtpengine-ctl set maxsessions -1' By default, the feature is disabled (i.e. maxsessions == -1).

  • --homer

    Enables sending the decoded contents of RTCP packets to a Homer SIP capture server. The transport is HEP version 3 and payload format is JSON. This argument takes an IP address and a port number as value.

  • --homer-protocol

    Can be either "udp" or "tcp" with "udp" being the default.

  • --homer-id

    The HEP protocol used by Homer contains a "capture ID" used to distinguish different sources of capture data. This ID can be specified using this argument.

  • --recording-dir

    An optional argument to specify a path to a directory where PCAP recording files and recording metadata files should be stored. If not specified, support for call recording will be disabled.

    Rtpengine supports multiple mechanisms for recording calls. See recording-method below for a list. The default recording method pcap is described in this section.

    PCAP files will be stored within a "pcap" subdirectory and metadata within a "metadata" subdirectory.

    The format for a metadata file is (with a trailing newline):

      /path/to/recording-pcap.pcap

  SDP mode: offer
  SDP before RTP packet: 1

  first SDP

  SDP mode: answer
  SDP before RTP packet: 1

  second SDP

  ...

  SDP mode: answer
  SDP before RTP packet: 100

  n-th and final SDP


  start timestamp (YYYY-MM-DDThh:mm:ss)
  end timestamp   (YYYY-MM-DDThh:mm:ss)


  generic metadata

    There are two empty lines between each logic block of metadata. We write out all answer SDP, each separated from one another by one empty line. The generic metadata at the end can be any length with any number of lines. Metadata files will appear in the subdirectory when the call completes. PCAP files will be written to the subdirectory as the call is being recorded.

    Since call recording via this method happens entirely in userspace, in-kernel packet forwarding cannot be used for calls that are currently being recorded and packet forwarding will thus be done in userspace only.

  • --recording-method

    Multiple methods of call recording are supported and this option can be used to select one. Currently supported are the method pcap and proc. The default method is pcap and is the one described above.

    The recording method proc works by writing metadata files directly into the recording-dir (i.e. not into a subdirectory) and instead of recording RTP packet data into pcap files, the packet data is exposed via a special interface in the /proc filesystem. Packets must then be retrieved from this interface by a dedicated 3rd party userspace component (usually a daemon).

    Packet data is held in kernel memory until retrieved by the userspace component, but only a limited number of packets (default 10) per media stream. If packets are not retrieved in time, they will be simply discarded. This makes it possible to flag all calls to be recorded and then leave it to the userspace component to decided whether to use the packet data for any purpose or not.

    In-kernel packet forwarding is fully supported with this recording method even for calls being recorded.

  • --recording-format

    When recording to pcap file in raw (default) format, there is no ethernet header. When set to eth, a fake ethernet header is added, making each package 14 bytes larger.

A typical command line (enabling both UDP and NG protocols) thus may look like:

/usr/sbin/rtpengine --table=0 --interface=10.64.73.31 --interface=2001:db8::4f3:3d \
--listen-udp=127.0.0.1:22222 --listen-ng=127.0.0.1:2223 --tos=184 \
--pidfile=/var/run/rtpengine.pid

In-kernel Packet Forwarding

In normal userspace-only operation, the overhead involved in processing each individual RTP or media packet is quite significant. This comes from the fact that each time a packet is received on a network interface, the packet must first traverse the stack of the kernel's network protocols, down to locating a process's file descriptor. At this point the linked user process (the daemon) has to be signalled that a new packet is available to be read, the process has to be scheduled to run, once running the process must read the packet, which means it must be copied from kernel space to user space, involving an expensive context switch. Once the packet has been processed by the daemon, it must be sent out again, reversing the whole process.

All this wouldn't be a big deal if it wasn't for the fact that RTP traffic generally consists of many small packets being tranferred at high rates. Since the forwarding overhead is incurred on a per-packet basis, the ratio of useful data processed to overhead drops dramatically.

For these reasons, rtpengine provides a kernel module to offload the bulk of the packet forwarding duties from user space to kernel space. Using this technique, a large percentage of the overhead can be eliminated, CPU usage greatly reduced and the number of concurrent calls possible to be handled increased.

In-kernel packet forwarding is implemented as an iptables module (or more precisely, an x_tables module). As such, it comes in two parts, both of which are required for proper operation. One part is the actual kernel module called xt_RTPENGINE. The second part is a plugin to the iptables and ip6tables command-line utilities to make it possible to actually add the required rule to the tables.

Overview

In short, the prerequisites for in-kernel packet forwarding are:

  1. The xt_RTPENGINE kernel module must be loaded.
  2. An iptables and/or ip6tables rule must be present in the INPUT chain to send packets to the RTPENGINE target. This rule should be limited to UDP packets, but otherwise there are no restrictions.
  3. The rtpengine daemon must be running.
  4. All of the above must be set up with the same forwarding table ID (see below).

The sequence of events for a newly established media stream is then:

  1. The SIP proxy (e.g. Kamailio) controls rtpengine and informs it about a newly established call.
  2. The rtpengine daemon allocates local UDP ports and sets up preliminary forward rules based on the info received from the SIP proxy. Only userspace forwarding is set up, nothing is pushed to the kernel module yet.
  3. An RTP packet is received on the local port.
  4. It traverses the iptables chains and gets passed to the xt_RTPENGINE module.
  5. The module doesn't recognize it as belonging to an established stream and thus ignores it.
  6. The packet continues normal processing and eventually ends up in the daemon's receive queue.
  7. The daemon reads it, processes it and forwards it. It also updates some internal data.
  8. This userspace-only processing and forwarding continues for a little while, during which time information about additional streams and/or endpoints may be obtained from the SIP proxy.
  9. After a few seconds, when the daemon is satisfied with what it has learned about the media endpoints, it pushes the forwarding rules to the kernel.
  10. From this moment on, the kernel module will recognize incoming packets belonging to those streams and will forward them on its own. It will stop those packets from traversing the network stacks any further, so the daemon will not see them any more on its receive queues.
  11. In-kernel forwarding is allowed to cease to work at any given time, either accidentally (e.g. by removal of the iptables rule) or deliberatly (the daemon will do so in case of a re-invite), in which case forwarding falls back to userspace-only operation.

The Kernel Module

The kernel module supports multiple forwarding tables (not to be confused with the tables managed by iptables), which are identified through their ID number. By default, up to 64 forwarding tables can be created and used, giving them the ID numbers 0 through 63.

Each forwarding table can be thought of a separate proxy instance. Each running instance of the rtpengine daemon controls one such table, and each table can only be controlled by one running instance of the daemon at any given time. In the most common setup, there will be only a single instance of the daemon running and there will be only a single forwarding table in use, with ID zero.

The kernel module can be loaded with the command modprobe xt_RTPENGINE. With the module loaded, a new directory will appear in /proc/, namely /proc/rtpengine/. After loading, the directory will contain only two pseudo-files, control and list. The control file is write-only and is used to create and delete forwarding tables, while the list file is read-only and will produce a list of currently active forwarding tables. With no tables active, it will produce an empty output.

The control pseudo-file supports two commands, add and del, each followed by the forwarding table ID number. To manually create a forwarding table with ID 42, the following command can be used:

echo 'add 42' > /proc/rtpengine/control

After this, the list pseudo-file will produce the single line 42 as output. This will also create a directory called 42 in /proc/rtpengine/, which contains additional pseudo-files to control this particular forwarding table.

To delete this forwarding table, the command del 42 can be issued like above. This will only work if no rtpengine daemon is currently running and controlling this table.

Each subdirectory /proc/rtpengine/$ID/ corresponding to each fowarding table contains the pseudo-files blist, control, list and status. The control file is write-only while the others are read-only. The control file will be kept open by the rtpengine daemon while it's running to issue updates to the forwarding rules during runtime. The daemon also reads the blist file on a regular basis, which produces a list of currently active forwarding rules together with their stats and other details within that table in a binary format. The same output, but in human-readable format, can be obtained by reading the list file. Lastly, the status file produces a short stats output for the forwarding table.

Manual creation of forwarding tables is normally not required as the daemon will do so itself, however deletion of tables may be required after shutdown of the daemon or before a restart to ensure that the daemon can create the table it wants to use.

The kernel module can be unloaded through rmmod xt_RTPENGINE, however this only works if no forwarding table currently exists and no iptables rule currently exists.

The iptables module

In order for the kernel module to be able to actually forward packets, an iptables rule must be set up to send packets into the module. Each such rule is associated with one forwarding table. In the simplest case, for forwarding table 42, this can be done through:

iptables -I INPUT -p udp -j RTPENGINE --id 42

If IPv6 traffic is expected, the same should be done using ip6tables.

It is possible but not strictly necessary to restrict the rules to the UDP port range used by rtpengine, e.g. by supplying a parameter like --dport 30000:40000. If the kernel module receives a packet that it doesn't recognize as belonging to an active media stream, it will simply ignore it and hand it back to the network stack for normal processing.

Summary

A typical start-up sequence including in-kernel forwarding might look like this:

# this only needs to be one once after system (re-) boot
modprobe xt_RTPENGINE
iptables -I INPUT -p udp -j RTPENGINE --id 0
ip6tables -I INPUT -p udp -j RTPENGINE --id 0

# ensure that the table we want to use doesn't exist - usually needed after a daemon
# restart, otherwise will error
echo 'del 0' > /proc/rtpengine/control

# start daemon
/usr/sbin/rtpengine --table=0 --ip=10.64.73.31 --ip6=2001:db8::4f3:3d \
--listen-ng=127.0.0.1:2223 --tos=184 --pidfile=/var/run/rtpengine.pid --no-fallback

Running Multiple Instances

In some cases it may be desired to run multiple instances of rtpengine on the same machine, for example if the host is multi-homed and has multiple usable network interfaces with different addresses. This is supported by running multiple instances of the daemon using different command-line options (different local addresses and different listening ports), together with multiple different kernel forwarding tables.

For example, if one local network interface has address 10.64.73.31 and another has address 192.168.65.73, then the start-up sequence might look like this:

modprobe xt_RTPENGINE
iptables -I INPUT -p udp -d 10.64.73.31 -j RTPENGINE --id 0
iptables -I INPUT -p udp -d 192.168.65.73 -j RTPENGINE --id 1

echo 'del 0' > /proc/rtpengine/control
echo 'del 1' > /proc/rtpengine/control

/usr/sbin/rtpengine --table=0 --ip=10.64.73.31 \
--listen-ng=127.0.0.1:2223 --tos=184 --pidfile=/var/run/rtpengine-10.pid --no-fallback
/usr/sbin/rtpengine --table=1 --ip=192.168.65.73 \
--listen-ng=127.0.0.1:2224 --tos=184 --pidfile=/var/run/rtpengine-192.pid --no-fallback

With this setup, the SIP proxy can choose which instance of rtpengine to talk to and thus which local interface to use by sending its control messages to either port 2223 or port 2224.

The ng Control Protocol

In order to enable several advanced features in rtpengine, a new advanced control protocol has been devised which passes the complete SDP body from the SIP proxy to the rtpengine daemon, has the body rewritten in the daemon, and then passed back to the SIP proxy to embed into the SIP message.

This control protocol is based on the bencode standard and runs over UDP transport. Bencoding supports a similar feature set as the more popular JSON encoding (dictionaries/hashes, lists/arrays, arbitrary byte strings) but offers some benefits over JSON encoding, e.g. simpler and more efficient encoding, less encoding overhead, deterministic encoding and faster encoding and decoding. A disadvantage over JSON is that it's not a readily human readable format.

Each message passed between the SIP proxy and the media proxy contains of two parts: a message cookie, and a bencoded dictionary, separated by a single space. The message cookie serves the same purpose as in the control protocol used by Kamailio's rtpproxy module: matching requests to responses, and retransmission detection. The message cookie in the response generated to a particular request therefore must be the same as in the request.

The dictionary of each request must contain at least one key called command. The corresponding value must be a string and determines the type of message. Currently the following commands are defined:

  • ping
  • offer
  • answer
  • delete
  • query
  • start recording

The response dictionary must contain at least one key called result. The value can be either ok or error. For the ping command, the additional value pong is allowed. If the result is error, then another key error-reason must be given, containing a string with a human-readable error message. No other keys should be present in the error case. If the result is ok, the optional key warning may be present, containing a human-readable warning message. This can be used for non-fatal errors.

For readabilty, all data objects below are represented in a JSON-like notation and without the message cookie. For example, a ping message and its corresponding pong reply would be written as:

{ "command": "ping" }
{ "result": "pong" }

While the actual messages as encoded on the wire, including the message cookie, might look like this:

5323_1 d7:command4:pinge
5323_1 d6:result4:ponge

All keys and values are case-sensitive unless specified otherwise. The requirement stipulated by the bencode standard that dictionary keys must be present in lexicographical order is not currently honoured.

The ng protocol is used by Kamailio's rtpengine module, which is based on the older module called rtpproxy-ng.

ping Message

The request dictionary contains no other keys and the reply dictionary also contains no other keys. The only valid value for result is pong.

offer Message

The request dictionary must contain at least the following keys:

  • sdp

    Contains the complete SDP body as string.

  • call-id

    The SIP call ID as string.

  • from-tag

    The SIP From tag as string.

Optionally included keys are:

  • via-branch

    The SIP Via branch as string. Used to additionally refine the matching logic between media streams and calls and call branches.

  • flags

    The value of the flags key is a list. The list contains zero or more of the following strings. Spaces in each string my be replaced by hyphens.

    • SIP source address

      Ignore any IP addresses given in the SDP body and use the source address of the received SIP message (given in received from) as default endpoint address. This was the default behaviour of older versions of rtpengine and can still be made the default behaviour through the --sip-source CLI switch. Can be overridden through the media address key.

    • trust address

      The opposite of SIP source address. This is the default behaviour unless the CLI switch --sip-source is active. Corresponds to the rtpproxy r flag. Can be overridden through the media address key.

    • symmetric

      Corresponds to the rtpproxy w flag. Not used by rtpengine as this is the default, unless asymmetric is specified.

    • asymmetric

      Corresponds to the rtpproxy a flag. Advertises an RTP endpoint which uses asymmetric RTP, which disables learning of endpoint addresses (see below).

    • unidirectional

      When this flag is present, kernelize also one-way rtp media.

    • strict source

      Normally, rtpengine attempts to learn the correct endpoint address for every stream during the first few seconds after signalling by observing the source address and port of incoming packets (unless asymmetric is specified). Afterwards, source address and port of incoming packets are normally ignored and packets are forwarded regardless of where they're coming from. With the strict source option set, rtpengine will continue to inspect the source address and port of incoming packets after the learning phase and compare them with the endpoint address that has been learned before. If there's a mismatch, the packet will be dropped and not forwarded.

    • media handover

      Similar to the strict source option, but instead of dropping packets when the source address or port don't match, the endpoint address will be re-learned and moved to the new address. This allows endpoint addresses to change on the fly without going through signalling again. Note that this opens a security hole and potentially allows RTP streams to be hijacked, either partly or in whole.

    • reset

      This causes rtpengine to un-learn certain aspects of the RTP endpoints involved, such as support for ICE or support for SRTP. For example, if ICE=force is given, then rtpengine will initially offer ICE to the remote endpoint. However, if a subsequent answer from that same endpoint indicates that it doesn't support ICE, then no more ICE offers will be made towards that endpoint, even if ICE=force is still specified. With the reset flag given, this aspect will be un-learned and rtpengine will again offer ICE to this endpoint. This flag is valid only in an offer message and is useful when the call has been transferred to a new endpoint without change of From or To tags.

    • port latching

      Forces rtpengine to retain its local ports during a signalling exchange even when the remote endpoint changes its port.

    • record call

      Identical to setting record call to on (see below).

  • replace

    Similar to the flags list. Controls which parts of the SDP body should be rewritten. Contains zero or more of:

    • origin

      Replace the address found in the origin (o=) line of the SDP body. Corresponds to rtpproxy o flag.

    • session connection or session-connection

      Replace the address found in the session-level connection (c=) line of the SDP body. Corresponds to rtpproxy c flag.

  • direction

    Contains a list of two strings and corresponds to the rtpproxy e and i flags. Each element must correspond to one of the named logical interfaces configured on the command line (through --interface). For example, if there is one logical interface named pub and another one named priv, then if side A (originator of the message) is considered to be on the private network and side B (destination of the message) on the public network, then that would be rendered within the dictionary as:

      { ..., "direction": [ "priv", "pub" ], ... }

    This only needs to be done for an initial offer; for the answer and any subsequent offers (between the same endpoints) rtpengine will remember the selected network interface.

    As a special case to support legacy usage of this option, if the given interface names are internal or external and if no such interfaces have been configured, then they're understood as selectors between IPv4 and IPv6 addresses. However, this mechanism for selecting the address family is now obsolete and the address family dictionary key should be used instead.

    A direction keyword is round-robin-calls. If this is received, a round robin algorithm runs for choosing the logical interface for the current stream(e.g. audio, video). The algorithm checks that all local interfaces of the tried logical interface have free ports for call streams. If a logical interface fails the check, the next one is tried. If there is no logical interface found with this property, it fallbacks to the default behaviour (e.g. return first logical interface in --interface list even if no free ports are available). The attribute is ignored for answers() because the logical interface was already selected at offers(). Naming an interface "round-robin-calls" and trying to select it using direction will run the above algorithm!

    Round robin for both legs of the stream: { ..., "direction": [ "round-robin-calls", "round-robin-calls" ], ... }

    Round robin for first leg and and select "pub" for the second leg of the stream: { ..., "direction": [ "round-robin-calls", "pub" ], ... }

    Round robin for first leg and and default behaviour for the second leg of the stream: { ..., "direction": [ "round-robin-calls" ], ... }

  • received from

    Contains a list of exactly two elements. The first element denotes the address family and the second element is the SIP message's source address itself. The address family can be one of IP4 or IP6. Used if SDP addresses are neither trusted (through SIP source address or --sip-source) nor the media address key is present.

  • ICE

    Contains a string, valid values are remove, force or force-relay. With remove, any ICE attributes are stripped from the SDP body. With force, ICE attributes are first stripped, then new attributes are generated and inserted, which leaves the media proxy as the only ICE candidate. The default behavior (no ICE key present at all) is: if no ICE attributes are present, a new set is generated and the media proxy lists itself as ICE candidate; otherwise, the media proxy inserts itself as a low-priority candidate.

    With force-relay, existing ICE candidates are left in place except relay type candidates, and rtpengine inserts itself as a relay candidate. It will also leave SDP c= and m= lines unchanged.

    This flag operates independently of the replace flags.

  • transport protocol

    The transport protocol specified in the SDP body is to be rewritten to the string value given here. The media proxy will expect to receive this protocol on the allocated ports, and will talk this protocol when sending packets out. Translation between different transport protocols will happen as necessary.

    Valid values are: RTP/AVP, RTP/AVPF, RTP/SAVP, RTP/SAVPF.

  • media address

    This can be used to override both the addresses present in the SDP body and the received from address. Contains either an IPv4 or an IPv6 address, expressed as a simple string. The format must be dotted-quad notation for IPv4 or RFC 5952 notation for IPv6. It's up to the RTP proxy to determine the address family type.

  • address family

    A string value of either IP4 or IP6 to select the primary address family in the substituted SDP body. The default is to auto-detect the address family if possible (if the recieving end is known already) or otherwise to leave it unchanged.

  • rtcp-mux

    A list of strings controlling the behaviour regarding rtcp-mux (multiplexing RTP and RTCP on a single port, RFC 5761). The default behaviour is to go along with the client's preference. The list can contain zero of more of the following strings. Note that some of them are mutually exclusive.

    • offer

      Instructs rtpengine to always offer rtcp-mux, even if the client itself doesn't offer it.

    • demux

      If the client is offering rtcp-mux, don't offer it to the other side, but accept it back to the offering client.

    • accept

      Instructs rtpengine to accept rtcp-mux and also offer it to the other side if it has been offered.

    • reject

      Reject rtcp-mux if it has been offered. Can be used together with offer to achieve the opposite effect of demux.

  • TOS

    Contains an integer. If present, changes the TOS value for the entire call, i.e. the TOS value used in outgoing RTP packets of all RTP streams in all directions. If a negative value is used, the previously used TOS value is left unchanged. If this key is not present or its value is too large (256 or more), then the TOS value is reverted to the default (as per --tos command line).

  • DTLS

    Contains a string and influences the behaviour of DTLS-SRTP. Possible values are:

    • off or no or disable

      Prevents rtpengine from offering or acceping DTLS-SRTP when otherwise it would. The default is to offer DTLS-SRTP when encryption is desired and to favour it over SDES when accepting an offer.

    • passive

      Instructs rtpengine to prefer the passive (i.e. server) role for the DTLS handshake. The default is to take the active (client) role if possible. This is useful in cases where the SRTP endpoint isn't able to receive or process the DTLS handshake packets, for example when it's behind NAT or needs to finish ICE processing first.

  • SDES

    A list of strings controlling the behaviour regarding SDES. The default is to offer SDES without any session parameters when encryption is desired, and to accept it when DTLS-SRTP is unavailable. If two SDES endpoints are connected to each other, then the default is to offer SDES with the same options as were received from the other endpoint.

    These options can also be put into the flags list using a prefix of SDES-. All options controlling SDES session parameters can be used either in all lower case or in all upper case.

    • off or no or disable

      Prevents rtpengine from offering SDES, leaving DTLS-SRTP as the other option.

    • unencrypted_srtp, unencrypted_srtcp and unauthenticated_srtp

      Enables the respective SDES session parameter (see section 6.3 or RFC 4568). The default is to copy these options from the offering client, or not to have them enabled if SDES wasn't offered.

    • encrypted_srtp, encrypted_srtcp and authenticated_srtp

      Negates the respective option. This is useful if one of the session parameters was offered by an SDES endpoint, but it should not be offered on the far side if this endpoint also speaks SDES.

  • record call

    Contains one of the strings yes, no, on or off. This tells the rtpengine whether or not to record the call to PCAP files. If the call is recorded, it will generate PCAP files for each stream and a metadata file for each call. Note that rtpengine will not force itself into the media path, and other flags like ICE=force may be necessary to ensure the call is recorded.

    See the --recording-dir option above.

    Enabling call recording via this option has the same effect as doing it separately via the start recording message, except that this option guarantees that the entirety of the call gets recorded, including all details such as SDP bodies passing through rtpengine.

  • metadata

    This is a generic metadata string. The metadata will be written to the bottom of metadata files within /path/to/recording_dir/metadata/. This can be used to record additional information about recorded calls. metadata values passed in through subsequent messages will overwrite previous metadata values.

    See the --recording-dir option above.

An example of a complete offer request dictionary could be (SDP body abbreviated):

{ "command": "offer", "call-id": "cfBXzDSZqhYNcXM", "from-tag": "mS9rSAn0Cr",
"sdp": "v=0\r\no=...", "via-branch": "5KiTRPZHH1nL6",
"flags": [ "trust address" ], "replace": [ "origin", "session connection" ],
"address family": "IP6", "received-from": [ "IP4", "10.65.31.43" ],
"ICE": "force", "transport protocol": "RTP/SAVPF", "media address": "2001:d8::6f24:65b",
"DTLS": "passive" }

The response message only contains the key sdp in addition to result, which contains the re-written SDP body that the SIP proxy should insert into the SIP message.

Example response:

{ "result": "ok", "sdp": "v=0\r\no=..." }

answer Message

The answer message is identical to the offer message, with the additional requirement that the dictionary must contain the key to-tag containing the SIP To tag. It doesn't make sense to include the direction key in the answer message.

The reply message is identical as in the offer reply.

delete Message

The delete message must contain at least the keys call-id and from-tag and may optionally include to-tag and via-branch, as defined above. It may also optionally include a key flags containing a list of zero or more strings. The following flags are defined:

  • fatal

    Specifies that any non-syntactical error encountered when deleting the stream (such as unknown call-ID) shall result in an error reply (i.e. "result": "error"). The default is to reply with a warning only (i.e. "result": "ok", "warning": ...).

Other optional keys are:

  • delete delay

    Contains an integer and overrides the global command-line option delete-delay. Call/branch will be deleted immediately if a zero is given. Value must be positive (in seconds) otherwise.

The reply message may contain additional keys with statistics about the deleted call. Those additional keys are the same as used in the query reply.

list Message

The list command retrieves the list of currently active call-ids. This list is limited to 32 elements by default.

  • limit

    Optional integer value that specifies the maximum number of results (default: 32). Must be > 0. Be careful when setting big values, as the response may not fit in a UDP packet, and therefore be invalid.

query Message

The minimum requirement is the presence of the call-id key. Keys from-tag and/or to-tag may optionally be specified.

The response dictionary contains the following keys:

  • created

    Contains an integer corresponding to the creation time of this call within the media proxy, expressed as seconds since the UNIX epoch.

  • last signal

    The last time a signalling event (offer, answer, etc) occurred. Also expressed as an integer UNIX timestamp.

  • tags

    Contains a dictionary. The keys of the dictionary are all the SIP tags (From-tag, To-Tag) known by rtpengine related to this call. One of the keys may be an empty string, which corresponds to one side of a dialogue which hasn't signalled its SIP tag yet. Each value of the dictionary is another dictionary with the following keys:

    • created

      UNIX timestamp of when this SIP tag was first seen by rtpengine.

    • tag

      Identical to the corresponding key of the tags dictionary. Provided to allow for easy traversing of the dictionary values without paying attention to the keys.

    • in dialogue with

      Contains the SIP tag of the other side of this dialogue. May be missing in case of a half-established dialogue, in which case the other side is represented by the null-string entry of the tags dictionary.

    • medias

      Contains a list of dictionaries, one for each SDP media stream known to rtpengine. The dictionaries contain the following keys:

      • index

        Integer, sequentially numbered index of the media, starting with one.

      • type

        Media type as string, usually audio or video.

      • protocol

        If the protocol is recognized by rtpengine, this string contains it. Usually RTP/AVP or RTP/SAVPF.

      • flags

        A list of strings containing various status flags. Contains zero of more of: initialized, rtcp-mux, DTLS-SRTP, SDES, passthrough, ICE.

      • streams

        Contains a list of dictionary representing the packet streams associated with this SDP media. Usually contains two entries, one for RTP and one for RTCP. The keys found in these dictionaries are listed below:

      • local port

        Integer representing the local UDP port. May be missing in case of an inactive stream.

      • endpoint

        Contains a dictionary with the keys family, address and port. Represents the endpoint address used for packet forwarding. The family may be one of IPv4 or IPv6.

      • advertised endpoint

        As above, but representing the endpoint address advertised in the SDP body.

      • crypto suite

        Contains a string such as AES_CM_128_HMAC_SHA1_80 representing the encryption in effect. Missing if no encryption is active.

      • last packet

        UNIX timestamp of when the last UDP packet was received on this port.

      • flags

        A list of strings with various internal flags. Contains zero or more of: RTP, RTCP, fallback RTCP, filled, confirmed, kernelized, no kernel support.

      • stats

        Contains a dictionary with the keys bytes, packets and errors. Statistics counters for this packet stream.

  • totals

    Contains a dictionary with two keys, RTP and RTCP, each one containing another dictionary identical to the stats dictionary described above.

A complete response message might look like this (formatted for readability):

      {
        "totals": {
          "RTCP": {
                "bytes": 2244,
                "errors": 0,
                "packets": 22
              },
          "RTP": {
               "bytes": 100287,
               "errors": 0,
               "packets": 705
             }
              },
        "last_signal": 1402064116,
        "tags": {
              "cs6kn1rloc": {
              "created": 1402064111,
              "medias": [
                      {
                  "flags": [
                         "initialized"
                       ],
                  "streams": [
                           {
                       "endpoint": {
                           "port": 57370,
                           "address": "10.xx.xx.xx",
                           "family": "IPv4"
                               },
                       "flags": [
                              "RTP",
                              "filled",
                              "confirmed",
                              "kernelized"
                            ],
                       "local port": 30018,
                       "last packet": 1402064124,
                       "stats": {
                              "packets": 343,
                              "errors": 0,
                              "bytes": 56950
                            },
                       "advertised endpoint": {
                                "family": "IPv4",
                                "port": 57370,
                                "address": "10.xx.xx.xx"
                              }
                           },
                           {
                       "stats": {
                              "bytes": 164,
                              "errors": 0,
                              "packets": 2
                            },
                       "advertised endpoint": {
                                "family": "IPv4",
                                "port": 57371,
                                "address": "10.xx.xx.xx"
                              },
                       "endpoint": {
                           "address": "10.xx.xx.xx",
                           "port": 57371,
                           "family": "IPv4"
                               },
                       "last packet": 1402064123,
                       "local port": 30019,
                       "flags": [
                              "RTCP",
                              "filled",
                              "confirmed",
                              "kernelized",
                              "no kernel support"
                            ]
                           }
                         ],
                  "protocol": "RTP/AVP",
                  "index": 1,
                  "type": "audio"
                      }
                    ],
              "in dialogue with": "0f0d2e18",
              "tag": "cs6kn1rloc"
                  },
              "0f0d2e18": {
                  "in dialogue with": "cs6kn1rloc",
                  "tag": "0f0d2e18",
                  "medias": [
                    {
                      "protocol": "RTP/SAVPF",
                      "index": 1,
                      "type": "audio",
                      "streams": [
                         {
                           "endpoint": {
                               "family": "IPv4",
                               "address": "10.xx.xx.xx",
                               "port": 58493
                             },
                           "crypto suite": "AES_CM_128_HMAC_SHA1_80",
                           "local port": 30016,
                           "last packet": 1402064124,
                           "flags": [
                            "RTP",
                            "filled",
                            "confirmed",
                            "kernelized"
                          ],
                           "stats": {
                            "bytes": 43337,
                            "errors": 0,
                            "packets": 362
                          },
                           "advertised endpoint": {
                              "address": "10.xx.xx.xx",
                              "port": 58493,
                              "family": "IPv4"
                            }
                         },
                         {
                           "local port": 30017,
                           "last packet": 1402064124,
                           "flags": [
                            "RTCP",
                            "filled",
                            "confirmed",
                            "kernelized",
                            "no kernel support"
                          ],
                           "endpoint": {
                               "family": "IPv4",
                               "port": 60193,
                               "address": "10.xx.xx.xx"
                             },
                           "crypto suite": "AES_CM_128_HMAC_SHA1_80",
                           "advertised endpoint": {
                              "family": "IPv4",
                              "port": 60193,
                              "address": "10.xx.xx.xx"
                            },
                           "stats": {
                            "packets": 20,
                            "bytes": 2080,
                            "errors": 0
                          }
                         }
                       ],
                      "flags": [
                       "initialized",
                       "DTLS-SRTP",
                       "ICE"
                     ]
                    }
                  ],
                  "created": 1402064111
                }
            },
        "created": 1402064111,
        "result": "ok"
      }

start recording Message

The start recording message must contain at least the key call-id and may optionally include from-tag, to-tag and via-branch, as defined above. The reply dictionary contains no additional keys.

Enables call recording for the call, either for the entire call or for only the specified call leg. Currently rtpengine always enables recording for the entire call and does not support recording only individual call legs, therefore all keys other than call-id are currently ignored.

If the chosen recording method doesn't support in-kernel packet forwarding, enabling call recording via this messages will force packet forwarding to happen in userspace only.