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.
- 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
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
andip6tables
. 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.
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 calledrtpengine
. 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
- libiptc library for iptables management (optional)
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.If you do not wish to (or cannot) compile the optional iptables management feature, the
Makefile
also contains a switch to disable it. See the--iptables-chain
option for a description. -
iptables-extension
Required for in-kernel packet forwarding.
With the
iptables
development headers installed, issuingmake
will compile the plugin foriptables
andip6tables
. The file will be calledlibxt_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 $VERSION is the output of the commanduname -r
. For example, if the commanduname -r
produces the output3.9-1-amd64
, then the kernel headers must be present in/lib/modules/3.9-1-amd64/build/
. The last component of this path (build
) is usually a symlink somewhere into/usr/src/
, which is fine.Successful compilation of the module will produce the file
xt_RTPENGINE.ko
. The module can be inserted into the running kernel manually throughinsmod xt_RTPENGINE.ko
(which will result in an error if depending modules aren't loaded, for example thex_tables
module), but it's recommended to copy the module into/lib/modules/$VERSION/updates/
, followed by runningdepmod -a
. After this, the module can be loaded by issuingmodprobe xt_RTPENGINE
.
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
--control-tos=INT TOS value to set on control-ng interface
-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
--redis-multikey Use multiple redis keys for storing the call (old behaviour) DEPRECATED
-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.
--iptables-chain=STRING Add explicit firewall rules to this iptables chain
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]
withIP
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 onepriv
by using--interface=pub/192.0.2.67 --interface=priv/10.35.2.75
. You can then use thedirection
option to tell rtpengine which local address to use for which endpoints (eitherpub
orpriv
).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 nameddefault
. You would use theaddress 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
andexternal
corresponding to thei
ande
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).
-
--control-tos
Takes an integer as argument and if given, specifies the TOS value that should be set in the control-ng interface packets. The default is to leave the TOS field untouched.
-
-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 example127.0.0.1:6379/12
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 examplefoobar@127.0.0.1:6379/12
. 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 variableRTPENGINE_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.
-
--redis-multikey
Use multiple redis keys for storing the call (old behaviour) DEPRECATED
-
-q, --no-redis-required When this parameter 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 methodpcap
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
andproc
. The default method ispcap
and is the one described above.The recording method
proc
works by writing metadata files directly into therecording-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.
-
--iptables-chain
This option enables explicit management of an iptables chain. When enabled, rtpengine takes control of the given iptables chain, which must exist already prior to starting the daemon. Upon startup, rtpengine will flush the chain, and then add one
ACCEPT
rule for each media port (RTP/RTCP) opened. Each rule will exactly match the individual port and destination IP address, and will be created with the call ID as iptables comment. The rule will be deleted when the port is closed.This option allows creating a firewall with a default
DROP
policy for the entire port range used by rtpengine and then referencing the given iptables chain to only selectively allow the ports actually in use.Note that this applies only to media ports, and does not apply to any other ports (such as the control ports) used by rtpengine.
Also note that the iptables API is not the most efficient one around and does not lend itself to fast dynamic creation and deletion of rules. If you have a high call volume, and especially many call attempts per second, you might experience significant performance impact. This is not a shortcoming of rtpengine but rather of iptables and its API implementation in the Linux kernel. In such a case, it is recommended to add a static iptables rule for the entire media port range instead, and not use this option.
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 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.
In short, the prerequisites for in-kernel packet forwarding are:
- The
xt_RTPENGINE
kernel module must be loaded. - An
iptables
and/orip6tables
rule must be present in theINPUT
chain to send packets to theRTPENGINE
target. This rule should be limited to UDP packets, but otherwise there are no restrictions. - The
rtpengine
daemon must be running. - 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:
- The SIP proxy (e.g. Kamailio) controls rtpengine and informs it about a newly established call.
- 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. - An RTP packet is received on the local port.
- It traverses the iptables chains and gets passed to the xt_RTPENGINE module.
- The module doesn't recognize it as belonging to an established stream and thus ignores it.
- The packet continues normal processing and eventually ends up in the daemon's receive queue.
- The daemon reads it, processes it and forwards it. It also updates some internal data.
- 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.
- 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.
- 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.
- 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 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 forwarding 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.
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.
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
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.
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
- stop 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 readability, 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.
The request dictionary contains no other keys and the reply dictionary also contains no other keys. The
only valid value for result
is pong
.
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. -
label
A custom free-form string which rtpengine remembers for this participating endpoint and reports back in logs and statistics output.
-
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 themedia 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 rtpproxyr
flag. Can be overridden through themedia address
key. -
symmetric
Corresponds to the rtpproxy
w
flag. Not used by rtpengine as this is the default, unlessasymmetric
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 thestrict 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 ifICE=force
is still specified. With thereset
flag given, this aspect will be un-learned and rtpengine will again offer ICE to this endpoint. This flag is valid only in anoffer
message and is useful when the call has been transferred to a new endpoint without change ofFrom
orTo
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
toon
(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
orsession-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
andi
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 namedpub
and another one namedpriv
, 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 theanswer
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
orexternal
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 theaddress 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
orIP6
. Used if SDP addresses are neither trusted (throughSIP source address
or--sip-source
) nor themedia address
key is present. -
ICE
Contains a string, valid values are
remove
,force
orforce-relay
. Withremove
, any ICE attributes are stripped from the SDP body. Withforce
, 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 (noICE
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 exceptrelay
type candidates, and rtpengine inserts itself as arelay
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
orIP6
to select the primary address family in the substituted SDP body. The default is to auto-detect the address family if possible (if the receiving 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.
-
require
Similar to
offer
but pretends that the receiving client has already accepted rtcp-mux. The effect is that no separate RTCP ports will be advertised, even in an initial offer (which is against RFC 5761). This option is provided to talk to WebRTC clients. -
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 ofdemux
.
-
-
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
orno
ordisable
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 ofSDES-
. All options controlling SDES session parameters can be used either in all lower case or in all upper case.-
off
orno
ordisable
Prevents rtpengine from offering SDES, leaving DTLS-SRTP as the other option.
-
unencrypted_srtp
,unencrypted_srtcp
andunauthenticated_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
andauthenticated_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
oroff
. 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 likeICE=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=..." }
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.
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.
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.
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. -
label
The label assigned to this endpoint in the
offer
oranswer
message. -
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
orvideo
. -
protocol
If the protocol is recognized by rtpengine, this string contains it. Usually
RTP/AVP
orRTP/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
andport
. Represents the endpoint address used for packet forwarding. Thefamily
may be one ofIPv4
orIPv6
. -
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
anderrors
. Statistics counters for this packet stream.
-
-
-
totals
Contains a dictionary with two keys,
RTP
andRTCP
, each one containing another dictionary identical to thestats
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"
}
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.
The stop recording
message must contain the key call-id
as defined above. The reply dictionary contains
no additional keys.
Disables call recording for the call. This can be sent during a call to imediatley stop recording it.