rtpengine/docs/ng_control_protocol.md

The NG Control Protocol

In order to provide 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. The SDP body gets rewritten by the daemon and then passed back to the SIP proxy in order to embed it into the SIP message.

This control protocol is supported over a number of different transports (plain UDP, plain TCP, HTTP, WebSocket) and loosely follows the same format as used by the module. Each message passed between the SIP and the media proxy consists of two parts separated by a single space:

• a unique message cookie
• a dictionary document

The message cookie is used to match requests to responses and to detect retransmissions. The message cookie in the response must be the same as in the request it's dedicated to.

The dictionary document can be in one of two formats:

• a JSON object
• a dictionary in bencode format

The bencoding mechanism supports a subset of JSON features, for example:

• dictionaries/hashes
• lists/arrays
• arbitrary byte strings

On the other hand, it offers some benefits over JSON encoding, e.g. simpler and more efficient encoding, less encoding overhead, deterministic encoding, faster encoding and decoding.

The disadvantages compared to JSON are that it's not readily a human readable format and sometimes it might be difficult to support it in programming languages.

Internally rtpengine uses the bencoding mechanism natively, leading to additional overhead when JSON is in use as it has to be converted.

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
• pause recording
• block DTMF
• unblock DTMF
• block media
• unblock media
• silence media
• unsilence media
• start forwarding
• stop forwarding
• play media
• stop media
• play DTMF
• statistics
• publish
• subscribe request
• subscribe answer
• unsubscribe
• connect

The response dictionary must contain at least one key called result. The value can be either ok or error.

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 the ping command, the additional value pong is allowed.

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

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

While the actual bencode encoded messages, 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 bencode standard's requirement that dictionary keys must be presented in the lexicographical order is currently not honored.

The NG protocol that is used by the module utilises the bencoding mechanism and the UDP transport by default, or, alternatively the websocket transport if enabled.

Of course the agent controlling rtpengine using the NG protocol does not have to be a SIP proxy (e.g. kamailio). Any process that involves SDP can potentially talk to rtpengine using this protocol.

As mentioned already, each NG-protocol message can include optional flags in order to cause specific behavior for this particular SDP offer/answer (e.g. transport, transcoding, preferred encryption parameters etc.)

The parsing of option flags (sometimes also called rtpp flags) can be done:

• by remote SIP proxy (e.g. kamailio)
• by rtpengine itself

*NOTE: currently parsing on the daemon side is implemented, but not all control agents may support it. As of the time of writing only the kamailio module uses it.

The difference between two approaches is that in the first case, the parsing of flags is done with help of module, meanwhile in the second case a list of flags is passed to rtpengine using bencode string format and is then parsed here. The benefit of the second approach is that any new flags supported by rtpengine will automatically be supported without having to worry about support in the control module.

When the flags are passed to rtpengine, they are formated as following:

{ "rtpp_flags": "replace-origin via-branch=auto-next strict-source label=callee OSRTP-accept transport-protocol=RTP/AVP address-family=IP4" }

Lists and dictionaries are supported in this format using square brackets [ ], for example:

{ "rtpp_flags": "via-branch=auto-next OSRTP=[accept] codec=[transcode=[PCMA PCMU] accept=[AMR-WB AMR] strip=[EVS]]" }

Regardless whether the flags parsing is done by the module or daemon, a functional behavior remains the same and has no difference in terms of SDP processing.

Messages description

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:

• all

  Can be set to the string none to disable any extra behaviour (which is the default if this key is omitted altogether) or to one of all, offer-answer, except-offer-answer or flows. Applicable to certain messages only. The behaviour is explained below separately for each affected message.

• 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 receiving end is known already) or otherwise to leave it unchanged.

• audio player

  Contains a string value of either default, transcoding, off, or always.

  The values transcoding and always result in the behaviour described under the audio-player config option in the manual, and override the global setting from the config file. The value off disables usage of the audio player regardless of the global config setting. The option default results in the behaviour mandated by the global config setting.

• delay-buffer

  Takes an integer as value. When set to non-zero, enables the delay buffer when setting up codec handlers. The delay buffer delays all media by the given number of milliseconds before passing it on. Once the delay buffer is configured, it must explicitly be disabled again by setting this value to zero. The delay buffer setting is honoured in all messages that set up codec handlers, such as block DTMF.

• 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.

  For legacy support, the special direction keyword round-robin-calls can be used to invoke the round-robin interface selection algorithm described in the section Interfaces configuration. If this special keyword is used, the round-robin selection will run over all configured interfaces, whether or not they are configured using the BASE:SUFFIX interface name notation. This special keyword is provided only for legacy support and should be considered obsolete. It will be removed in future versions.

  For commands that require only one interface (e.g. publish), use the interface=... key. For commands that require two interfaces, as an alternative to the direction= key, the two interfaces can be listed separately, using from-interface=... for the first interface and to-interface=... for the second one.

• digit or code

  Sets the replacement digit for DTMF-security=DTMF.

• drop-traffic

  Contains a string, valid values are start or stop.

  start signals to rtpengine that all RTP involved in this call is dropped. Can be present either in offer or answer, the behavior is for the entire call.

  stop signals to rtpengine that all RTP involved in this call is NOT dropped anymore. Can be present either in offer or answer, the behavior is for the entire call.

  stop has priority over start, if both are present.

• 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.

  • active

    Reverts the passive setting. Only useful if the dtls-passive config option is set.

• DTLS-reverse

  Contains a string and influences the behaviour of DTLS-SRTP. Unlike the regular DTLS flag, this one is used to control behaviour towards DTLS that was offered to rtpengine. In particular, if passive mode is used, it prevents rtpengine from prematurely sending active DTLS connection attempts. Possible values are:

  • 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.

  • active

    Reverts the passive setting. Only useful if the dtls-passive config option is set.

• DTLS-fingerprint

  Contains a string and is used to select the hashing function to generate the DTLS fingerprint from the certificate. The default is SHA-256, or the same hashing function as was used by the peer. Available are SHA-1, SHA-224, SHA-256, SHA-384, and SHA-512.

• DTMF-security

  Used in the block DTMF message to select the DTMF blocking mode. The default mode is drop which simply drops DTMF event packets. The other supported modes are: silence which replaces DTMF events with silence audio; tone which replaces DTMF events with a single sine wave tone; random which replaces DTMF events with random other DTMF events (both in-band DTMF audio tones and RFC event packets); zero which is similar to random except that a zero event is always used; DTMF which is similar to zero except that a different DTMF digit can be specified; off to disable DTMF blocking.

• DTMF-security-trigger

  Blocking mode to enable when the DTMF trigger (see below) is detected.

• DTMF-security-trigger-end

  Blocking mode to enable when the DTMF end trigger (see below) is detected.

• DTMF-delay

  Time in milliseconds to delay DTMF events (both RFC event packets and DTMF tones) for. With this option enabled (set to non-zero), DTMF events are initially replaced by silence and then subsequently reproduced after the given delay. DTMF blocking modes are honoured at the time when the DTMF events are reproduced.

• DTMF-log-dest

  Contains a destination address and port for the DTMF logging feature. This overrides the global destination from the dtmf-log-dest config option on a per-call basis. Even if the global config option is unset, setting the destination address/port via this option enables DTMF logging for this call.

• endpoint-learning

  Contains one of the strings off, immediate, delayed or heuristic. This tells rtpengine which endpoint learning algorithm to use and overrides the endpoint-learning configuration option. This option can also be put into the flags list using a prefix of endpoint-learning-.

• from-interface

  Contains a string identifying the network interface pertaining to the "received from" direction of this message. Identical to setting the first direction= value.

• frequency or frequencies

  Sets the tone frequency or frequencies for DTMF-security=tone in Hertz. The default is a single frequency of 400 Hz. A list of frequencies can be given either as a list object, or as a string containing a comma-separated list of integers. The given frequencies will be picked from the list in order, one for each DTMF event detected, and will be repeated once the end of the list is reached.

• from-tags

  Contains a list of strings used to selected multiple existing call participants (e.g. for the subscribe request message). An alternative way to list multiple tags is by putting them into the flags list, each prefixed with from-tags-.

• generate RTCP

  Contains a string, either on or off. If enabled for a call, received RTCP packets will not simply be passed through as usual, but instead will be consumed, and instead rtpengine will generate its own RTCP packets to send to the RTP peers. This flag will be effective for both sides of a call.

• ICE

  Contains a string which must be one of the following values:

  With remove, any ICE attributes are stripped from the SDP body. Also see the flag reject ICE to effect an early removal of ICE support during an offer.

  With force, ICE attributes are first stripped, then new attributes are generated and inserted, which leaves the media proxy as the only ICE candidate.

  With default, the behaviour will be the same as with force if the incoming SDP already had ICE attributes listed. If the incoming SDP did not contain ICE attributes, then no ICE attributes are added.

  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.

  With optional, 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. This used to be the default behaviour in previous versions of rtpengine.

  The default behaviour (no ICE key present at all) is the same as default.

  This flag operates independently of the replace flags.

  Note that if config parameter save-interface-ports = true, ICE will be broken, because rtpengine will bind ports only on the first local interface of desired family of logical interface.

• ICE-lite

  Contains a string which must be one of the following values:

  • forward to enable "ICE lite" mode towards the peer that this offer is sent to.

  • backward to enable "ICE lite" mode towards the peer that has sent this offer.

  • both to enable "ICE lite" towards both peers.

  • off to disable "ICE lite" towards both peers and revert to full ICE support.

  The default (keyword not present at all) is to use full ICE support, or to leave the previously set "ICE lite" mode unchanged. This keyword is valid in offer messages only.

• interface

  Contains a single string naming one of the configured interfaces, just like direction does. The interface option is used instead of direction where only one interface is required (e.g. outside of an offer/answer scenario), for example in the publish or subscribe request commands.

• label or from-label

  A custom free-form string which rtpengine remembers for this participating endpoint and reports back in logs and statistics output. For some commands (e.g. block media) the given label is not used to set the label of the call participant, but rather to select an existing call participant.

• 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.

• media echo or media-echo

  Contains a string to enable a special media echo mode. Recognised values are:

  • blackhole or sinkhole

    Media arriving from either side of the call is simply discarded and not forwarded.

  • forward

    Enables media echo towards the receiver of this message (e.g. the called party if the message is an offer from the caller). Media arriving from that side is echoed back to its sender (with a new SSRC if it's RTP). Media arriving from the opposite side is discarded.

  • backwards

    Enables media echo towards the sender of this message (i.e. the opposite of forward). Media arriving from the other side is discarded.

  • both

    Enables media echo towards both the sender and the receiver of this message.

• metadata

  This is a generic metadata string. The metadata will be written to the bottom of metadata files within /path/to/recording_dir/metadata/ or to recording_metakeys table. In the latter case, metadata string must contain a list of key:val pairs separated by | character. metadata 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.

• OSRTP

  Similar to SDES but controls OSRTP behaviour. Default behaviour is to pass through OSRTP negotiations. Supported options:

  • offer or offer-RFC

    When processing a non-OSRTP offer, convert it to an OSRTP offer. Will result in RTP/SRTP transcoding if the OSRTP offer is accepted. The transport protocol should be a non-SRTP (plain RTP) protocol such as RTP/AVP.

  • offer-legacy

    Convert a regular offer to a legacy, non-RFC "best effort" SRTP offer, which involves duplicating each SDP media section in the output, advertised once as plain RTP and once as SRTP. The transport protocol should be set to an SRTP protocol such as RTP/SAVP. To enable full interoperability with endpoints which support this usage, the flag accept-legacy (see below) should also be given in all signalling exchanges.

  • accept-RFC

    When processing a non-OSRTP answer in response to an OSRTP offer, accept the OSRTP offer anyway. Results in RTP/SRTP transcoding.

  • accept-legacy

    Enables support for legacy, non-RFC "best effort" SRTP offers, which consist of media sections being advertised twice, once as plain RTP and once as SRTP. With this option set, rtpengine will treat such SDPs as SRTP SDPs, removing the duplicated media sections. This flag must be given for both offer and answer messages.

  • accept

    Short for both accept-RFC and accept-legacy. Can be used unconditionally in all signalling if so desired.

• output-destination

  See start recording below.

• ptime

  Contains an integer. If set, changes the a=ptime attribute's value in the outgoing SDP to the provided value. It also engages the transcoding engine for supported codecs to provide repacketization functionality, even if no additional codec has actually been requested for transcoding. Note that not all codecs support all packetization intervals.

  The selected ptime (which represents the duration of a single media packet in milliseconds) will be used towards the endpoint receiving this offer, even if the matching answer prefers a different ptime.

  This option is ignored in answer messages. See below for the reverse.

• ptime-reverse

  This is the reciprocal to ptime. It sets the ptime to be used towards the endpoint who has sent the offer. It will be inserted in the answer SDP. This option is also ignored in answer messages.

• 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.

• record call

  Contains one of the strings yes, no, on or off. This tells 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.

• 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 of demux.

• SIP message type

  Contains a string indicating whether the SIP message that triggered this signalling message was either a SIP request or a SIP response.

• SIP code

  Contains an integer corresponding to the SIP response code (e.g. 180 or 200) if this signalling message was triggered by a SIP response.

• template

  Contains the name of a signalling template to be used for this particular control message. See documentation for SIGNALLING TEMPLATES in the man page.

• via-branch

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

• set-label

  Some commands (e.g. block media) use the given label to select an existing call participant. For these commands, set-label instead of label can be used to set the label at the same time, either for the selected call participant (if selected via from-tag) or for the newly created participant (e.g. for subscribe request).

• 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. Additionally, all other supported SDES crypto suites are added to the outgoing offer by default.

  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.

  • no-SUITE

    Exclude individual crypto suites from being included in the offer. For example, no-NULL_HMAC_SHA1_32 would exclude the crypto suite NULL_HMAC_SHA1_32 from the offer. This has two effects: if a given crypto suite was present in a received offer, it will be removed and will be missing in the outgoing offer; and if a given crypto suite was not present in the received offer, it will not be added to it.

    Remark: if after applying the policies to the processed offer, there are no crypto suites left, which can be used later in the answer towards the offerer, then rtpengine will intentionally leave the top most one offered, for the answer towards the originator. However it will be not used for the recipient.

  • only-SUITE

    Add only these individual crypto suites and none of the others. For example, only-NULL_HMAC_SHA1_32 would only accept the crypto suite NULL_HMAC_SHA1_32 for the offer being generated. This takes precedence over the SDES-no- flag(s), if used together, so the SDES-no will be not taken into account. This has two effects: if a given crypto suite was present in a received offer, it will be kept, so will be present in the outgoing offer; and if a given crypto suite was not present in the received offer, it will be added to it. The rest, which is not mentioned, will be dropped/not added.

    Remark: if after applying the policies to the processed offer, there are no crypto suites left, which can be used later in the answer towards the offerer, then rtpengine will intentionally leave the top most one offered, for the answer towards the originator. However it will be not used for the recipient.

  • nonew

    Don't add any new crypto suites into the offer. This means, offered SDES crypto suites will be accepted, meanwhile no new are going to be generated by rtpengine. It takes precedence over the SDES-no and SDES-only flags, if used in combination.

  • order:SUITES LIST

    The order, in which crypto suites are being added to the SDP. Example: SDES-order:AES_256_CM_HMAC_SHA;AES_256_CM_HMAC_SHA1_32;AES_192_CM_HMAC_SHA1_80;, this means — those listed SDES crypto suites will be added into the generated SDP body at the top of crypto suites list, in the given order. But, each of them is added, only if it is about to be added/generated. In other words, the SDES-order: flag itself doesn't add crypto suites, it just affects the order of those suites to be added.

    And the rest of non-mentioned suites (not mentioned in the SDES-order: list), which are also to be added, will be appended after those given, in the free manner of ordering.

    Important thing to remember - it doesn't change the crypto suite tag for the recipient, even though changing the order of them.

    This flag does not contradict with SDES-nonew, SDES-only- and SDES-no- flags. It just orders the list of crypto suites already prepared to be sent out.

  • offerer_pref:SUITES LIST

    The list of preferred crypto suites to be selected for the offerer.

    It provides a possibility to select specific crypto suite(s) for the offerer from the given list of crypto suites received in the offer.

    This will be used later on, when processing an answer from the recipient and generating an answer to be sent out towards offerer.

    Furthermore, this is being decided not when the answer is processed, but already when the offer is processed.

    Flag usage example: SDES-offerer_pref:AES_256_CM_HMAC_SHA;AES_256_CM_HMAC_SHA1_32;

  • pad

    RFC 4568 (section 6.1) is somewhat ambiguous regarding the base64 encoding format of a=crypto parameters added to an SDP body. The default interpretation is that trailing = characters used for padding should be omitted. With this flag set, these padding characters will be left in place.

  • lifetime

    Add the key lifetime parameter 2^31 to each crypto key.

  • static

    Instructs rtpengine to skip the full SDES negotiation routine during a re-invite (e.g. pick the first support crypto suite, look for possible SRTP passthrough) and instead leave the previously negotiated crypto suite in place. Only useful in subsequent answer messages and ignored in offer messages.

  • prefer

    If an offer or publish contain both DTLS and SDES options, by default rtpengine prefers DTLS over SDES and would end up accepting DTLS. With this option set, in this scenario SDES would be preferred and accepted, while DTLS would be rejected. Useful in combination with DTLS=off.

• supports

  Contains a list of strings. Each string indicates support for an additional feature that the controlling SIP proxy supports. Currently defined values are:

  • load limit

    Indicates support for an extension to the ng protocol to facilitate certain load balancing mechanisms. If rtpengine is configured with certain session or load limit options enabled (such as the max-sessions option), then normally rtpengine would reply with an error to an offer if one of the limits is exceeded. If support for the load limit extension is indicated, then instead of replying with an error, rtpengine responds with the string load limit in the result key of the response dictionary. The response dictionary may also contain the optional key message with an explanatory string. No other key is required in the response dictionary.

• to-interface

  Contains a string identifying the network interface pertaining to the "going to" direction of this message. Identical to setting the second direction= value.

• to-label

  Commands that allow selection of two call participants (e.g. block media) can use label instead of from-tag to select the first call participant. The to-label can then be used instead of to-tag to select the other call participant.

  For subscribe request the to-label is synonymous with set-label.

• 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).

• 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.

  Additionally the string accept can be given in answer messages to allow a special case: By default (when no transport-protocol override is given) in answer messages, rtpengine will use the transport protocol that was originally offered. However, an answering client may answer with a different protocol than what was offered (e.g. offer was for RTP/AVP and answer comes with RTP/AVPF). The default behaviour for rtpengine is to ignore this protocol change and still proceed with the protocol that was originally offered. Using the accept option here tells rtpengine to go along with this protocol change and pass it to the original offerer.

• trigger

  A string of DTMF digits that enable a DTMF blocking mode when detected.

• trigger-end or end trigger

  A string of DTMF digits that disable DTMF blocking or enable a different DTMF blocking mode when detected, but only after the initial enabling trigger has been detected.

• trigger-end-time

  Time in milliseconds that a DTMF blocking mode enabled by the trigger should remain active the most. After the time has expired, the blocking mode is switched to the trigger-end mode.

• trigger-end-digits

  Number of DTMF digits that a DTMF blocking mode enabled by the trigger should remain active the most. After this number of DTMF digits has been detected, the blocking mode is switched to the trigger-end mode.

• T.38

  Contains a list of strings. Each string is a flag that controls the behaviour regarding T.38 transcoding. These flags are ignored if the message is not an offer. Recognised flags are:

  • decode

    If the received SDP contains a media section with an image type, UDPTL transport, and t38 format string, this flag instructs rtpengine to convert this media section into an audio type using RTP as transport protocol. Other transport protocols (such as SRTP) can be selected using transport protocol as described above.

    The default audio codecs to be offered are PCMU and PCMA. Other audio codecs can be specified using the transcode= flag described above, in which case the default codecs will not be offered automatically.

  • force

    If the received SDP contains an audio media section using RTP transport, this flag instructs rtpengine to convert it to an image type media section using the UDPTL protocol. The first supported audio codec that was offered will be used to transport T.30. Default options for T.38 are used for the generated SDP.

  • stop

    Stops a currently active T.38 gateway that was previously engaged using the decode or force flags. This is useful to handle a rejected T.38 offer and revert the session back to media passthrough.

  • no-ECM

    Disable support for ECM. Support is enabled by default.

  • no-V.17

    Disable support for V.17. Support is enabled by default.

  • no-V.27ter

    Disable support for V.27ter. Support is enabled by default.

  • no-V.29

    Disable support for V.29. Support is enabled by default.

  • no-V.34

    Disable support for V.34. Support is enabled by default.

  • no-IAF

    Disable support for IAF. Support is enabled by default.

  • FEC

    Use UDPTL FEC instead of redundancy. Only useful with T.38=force as it's a negotiated parameter.

• T.38 version

  Sets the T.38 version number to use for the T.38 gateway. The default is version zero, or to go along with what has been advertised in the SDP if responding to a received T.38 offer. Overriding the version to zero regardless of what has been advertised in the SDP can solve T.38 gateway problems against certain endpoints.

• volume

  Sets the tone volume for DTMF-security modes tone, zero, DTMF, and random` in negative dB. The default is -10 dB. The highest possible volume is 0 dB and the lowest possible volume is -63 dB.

• xmlrpc-callback

  Contains a string that encodes an IP address (either IPv4 or IPv6) in printable format. If specified, then this address will be used as destination address for the XMLRPC timeout callback (see b2b-url option).

Optionally included flags are:

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

• all

  Synonymous to all=all (see below).

• allow asymmetric codecs

  Normally rtpengine expects codecs that were offered during an SDP offer to match the ones that are accepted in the corresponding SDP answer. This expectation includes the RTP payload type number. In particular this is relevant to codecs using dynamic RTP payload type numbering (generally 96 and above). For example if the SDP offer included AMR-WB with payload type number 98, then the answering client is expected to also use payload type number 98 if it wanted to accept this codec.

  With this option set, mismatched payload type numbers are accepted and honoured. If an answering client accepts a codec that was not offered (with that payload type number), then a lookup is performed in attempt to find a matching and compatible codec from the offer with a different payload type number. If a match is found then the codec is considered as accepted.

  Note that payload type number translation will not be performed in this situation.

• allow no codec media

  Enables special handling for SDP media sections (m= lines) that are left without any codecs after codec manipulation operations (in particular codec stripping) have been performed. By default without this option set, a media section without any codecs would be considered a usage error, and the original list of codecs would be restored so that media flow can be established. With this option set, a media section without any codecs would be considered intentionally so, and would be converted to a rejected or removed media section, that is a media section with a zero port, a dummy format list, and further attributes.

• allow transcoding

  This flag is only useful in commands that provide an explicit answer SDP to rtpengine (e.g. subscribe answer). For these commands, if the answer SDP does not accept all codecs that were offered, the default behaviour is to reject the answer. With this flag given, the answer will be accepted even if some codecs were rejected, and codecs will be transcoded as required.

• force transcoding

  This flag will force transcoding between channels. This provides a loss measurement between the A-leg and B-leg. If this flag appears in the NG protocol, it will always be transcoded (for example, between codecs G722 and G722).

• always transcode

  Legacy flag, synonymous to codec-accept=all.

• asymmetric

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

• block DTMF

  Useful in offer or answer messages to immdiately enable DTMF blocking (or other DTMF security mechanism) for the relevant call party, identical to using a block DTMF message for the call party immediately after.

• block egress

  Instructs rtpengine to suppress and block other egress media to a remote client while media playback towards that client is ongoing. Useful for play media messages, as well as offer and answer in combination with recording announcement.

• block short or block short packets

  Enables blocking of short RTP packets for the applicable call participant. Short RTP packets are packets shorter than the expected minimum length, which is determined empirically based on what is observed on the wire. Short packets are simply discarded. This is supported only for codecs for which a fixed packet size is expected (e.g. G.711).

• debug or debugging

  Enabled full debug logging for this call, regardless of global log level settings.

• detect DTMF

  When present in a message that sets up codec handlers, enables the DSP to detect in-band DTMF audio tones even when it wouldn't otherwise be necessary.

• discard recording

  When file recording is in use, instructs the recording daemon to discard (delete) the recording files, as well as the database entries if present.

• exclude recording

  Instructs rtpengine to exclude this call participant's media from being recorded. When used within an offer/answer exchange, applies to both call parties involved.

• skip-recording-db

  Suppress writing the information about the call recording to the configured metadata DB.

• early media

  Used in conjunction with the audio player. If set, audio playback is started immediately when processing an offer message. The default behaviour is to start the audio player only after the answer has been processed, or when any audio to be played back has actually been received (either from another party to the call, or via the play media command).

• full rtcp attribute

  Include the full version of the a=rtcp line (complete with network address) instead of the short version with just the port number.

• generate RTCP

  Identical to setting generate RTCP = on.

• generate mid

  Add a=mid attributes to the outgoing SDP if they were not already present.

• inactive

  Useful for subscribe request messages to produce an SDP which is marked as inactive, instead of sendonly which is the default. This can be used to pause media sent to a subscription.

• inject DTMF

  Signals to rtpengine that the audio streams involved in this offer or answer (the flag should be present in both of them) are to be made available for DTMF injection via the play DTMF control message. See play DTMF below for additional information.

• loop protect

  Inserts a custom attribute (a=rtpengine:...) into the outgoing SDP to prevent rtpengine processing and rewriting the same SDP multiple times. This is useful if your setup involves signalling loops and need to make sure that rtpengine doesn't start looping media packets back to itself. When this flag is present and rtpengine sees a matching attribute already present in the SDP, it will leave the SDP untouched and not process the message.

• 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.

• NAT-wait

  Prevents forwarding media packets to the respective endpoint until at least one media packet has been received from that endpoint. This is to allow a NAT binding to open in the ingress direction before sending packets out, which could result in an automated firewall block.

• new branch

  If rtpengine receives an answer from a to-tag that hasn't previously seen and no corresponding call party is known (created from a previous offer), previously it would treat this is a separate new call branch, create a brand new internal call party, and dissociate the previous one. This may lead to unexpected results as this new call party has been created without the same initialisation as was done for the original one, and so may be left with incorrect or incomplete data (e.g. SRTP keys, codec information, interface bindings, etc).

  Improve this by treating an unexpected and unseen to-tag as an alias to the already existing to-tag. Going forward both tags can then be used interchangeably to refer to the same monologue.

  This flag suppresses this new behaviour, in case some situation is made worse by it.

• no port latching

  Port latching is enabled by default for endpoints which speak ICE. With this option preset, a remote port change will result in a local port change even for endpoints which speak ICE, which will imply an ICE restart.

• no rtcp attribute

  Omit the a=rtcp line from the outgoing SDP.

• original sendrecv

  With this flag present, rtpengine will leave the media direction attributes (sendrecv, recvonly, sendonly, and inactive) from the received SDP body unchanged. Normally rtpengine would consume these attributes and insert its own version of them based on other media parameters (e.g. a media section with a zero IP address would come out as sendonly or inactive).

• pad crypto

  Legacy alias to SDES=pad.

• pierce NAT

  Sends empty UDP packets to the remote RTP peer as soon as an endpoint address is available from a received SDP, for as long as no incoming packets have been received. Useful to create an initial NAT mapping. Not needed when ICE is in use.

• port latching

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

• provisional

  Disables special behaviour when operating on a message that was triggered by a SIP response with a non-provisional (>= 200) status code. Specifically, setting this flag allows for changed or updated to-tag even after a final SIP response has been received.

• record call

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

• recording announcement

  Enable playback of an announcement message when call recording is started. One of the flags identifying a media file (such as file=, same as for the play media message) must also be given, and generally usage of block egress is recommended.

  Announcement messages are enabled directionally, meaning this flag enables it for the call party relevant to the current message (e.g the call originator for an initial invite) but not for other. In other words this flag must be set for all call parties which are meant to hear the announcement.

• reject ICE

  Useful for offer messages that advertise support for ICE. Instructs rtpengine to reject the offered ICE. This is similar to using ICE=remove in the respective answer.

• 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.

• reuse codecs or no codec renegotiation

  Instructs rtpengine to prevent endpoints from switching codecs during call run-time if possible. Codecs that were listed as preferred in the past will be kept as preferred even if the re-offer lists other codecs as preferred, or in a different order. Recommended to be combined with single codec.

• RTCP mirror

  Useful only for subscribe request message. Instructs rtpengine to not only create a one-way subscription for both RTP and RTCP from the source to the sink, but also create a reverse subscription for RTCP only from the sink back to the source. This makes it possible for the media source to receive feedback from all media receivers (sinks).

• single codec

  Using this flag in an answer message will leave only the first listed codec in place and will remove all others from the list. Useful for RTP clients which get confused if more than one codec is listed in an answer.

• static codecs

  Useful in an offer message to suppress any change in codecs towards the answer side, instead of passing along the list of offered codecs from the offer side as it normally would.

• 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.

• symmetric

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

• 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.

• strip extmap

  Remove a=rtpmap attributes from the outgoing SDP.

• 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.

• trickle ICE

  Useful for offer messages when ICE is advertised to also advertise support for trickle ICE.

• unidirectional

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

• WebRTC

  Shortcut alias for several other flags that must be set when talking to a WebRTC client. Currently an alias for (subject to change): transport-protocol=UDP/TLS/RTP/SAVPF ICE=force tricke-ICE rtcp-mux-offer rtcp-mux-require no-rtcp-attribute SDES-off generate-mid

Optionally included replace-flags are:

Similar to the usual flags list, but this one controls which parts of the SDP body should be rewritten. Contains zero or more of:

• force-increment-sdp-ver

  Force increasing the SDP version, even if the SDP hasn't been changed.

• origin

  Replace the address found in the origin (o=) line of the SDP body.

• origin-full

  Replace whole origin (o=) line of the SDP body, so that all origin fields in the o= line always remain the same in all SDPs going to a particular RTP endpoint. A behavior in relation to the address field is the same as by the origin option flag.

• session name or session-name

  Same as username but for the entire contents of the s= line.

• SDP version or SDP-version

  Take control of the version field in the SDP and make sure it's increased every time the SDP changes, and left unchanged if the SDP is the same.

• username

  Take control of the origin username field in the SDP. With this option in use, rtpengine will make sure the username field in the o= line always remains the same in all SDPs going to a particular RTP endpoint.

• zero address

  Using a zero endpoint address is an obsolete way to signal a muted or sendonly stream. Streams with zero addresses are normally flagged as sendonly and the zero address in the SDP is passed through. With this option set, the zero address is replaced with a real address.

Optionally included codec manipulations:

codec contains a dictionary controlling various aspects of codecs (or RTP payload types).

These options can also be put into the flags list using a prefix of codec-. For example, to set the codec options for two variants of Opus when they're implicitly accepted, (see the example under set), one would put the following into the flags list: codec-set-opus/48000/1/16000 codec-set-opus/48000/2/32000

The following keys are understood:

• accept

  Similar to mask and consume but doesn't remove the codec from the list of offered codecs. This means that a codec listed under accept will still be offered to the remote peer, but if the remote peer rejects it, it will still be accepted towards the original offerer and then used for transcoding. It is a more selective version of what the always transcode flag does.

  The special string any can be used for the publish message. See below for more details.

• consume

  Identical to mask but enables the transcoding engine even if no other transcoding related options are given.

• except

  Contains a list of strings. Each string is the name of a codec that should be included in the list of codecs offered. This is primarily useful to block all codecs (strip -> all or mask -> all) except the ones given in the except whitelist. Codecs that were not present in the original list of codecs offered by the client will be ignored.

  This list also supports codec format parameters as per above.

• ignore

  Similar to the strip option below, but affects only codecs listed in the incoming received SDP. Codecs listed here are treated as if they were never offered, and so will not be used for media towards the offerer. Note that codecs listed here would still be used in the outgoing rewritten offer SDP, unless the same codecs are also listed under strip. This means that if a codec is only ignored but not stripped, and if that codec is then accepted by the answerer, transcoding will necessarily be enabled.

• mask

  Similar to strip except that codecs listed here will still be accepted and used for transcoding on the offering side. Useful only in combination with transcode. For example, if an offer advertises Opus and the options mask=opus, transcode=G723 are given, then the rewritten outgoing offer will contain only G.723 as offered codec, and transcoding will happen between Opus and G.723. In contrast, if only transcode=G723 were given, then the rewritten outgoing offer would contain both Opus and G.723. On the other hand, if strip=opus, transcode=G723 were given, then Opus would be unavailable for transcoding.

  As with the strip option, the special keywords all and full can be used to mask all codecs that have been offered.

  This option is only processed in offer messages and ignored otherwise.

• offer

  This is identical to except but additionally allows the codec order to be changed. So the first codec listed in offer will be the primary (preferred) codec in the output SDP, even if it wasn't originally so.

• set

  Contains a list of strings. This list makes it possible to set codec options (bitrate in particular) for codecs that are implicitly accepted for transcoding. For example, if AMR was offered, transcode=PCMU was given, and the remote ended up accepting PCMU, then this option can be used to set the bitrate used for the AMR transcoding process.

  Each string must be a full codec specification as per above, including clock rate and number of channels. Using the example above, set=AMR/8000/1/7400 can be used to transcode to AMR with 7.4 kbit/s.

  Codec options (bitrate) are only applied to codecs that match the given parameters (clock rate, channels), and multiple options can be given for the same coded with different parameters. For example, to specify different bitrates for Opus for both mono and stereo output, one could use set=[opus/48000/1/16000,opus/48000/2/32000].

  This option is only processed in offer messages and ignored otherwise.

• strip

  Contains a list of strings. Each string is the name of a codec or RTP payload type that should be removed from the SDP. Codec names are case sensitive, and can be either from the list of codecs explicitly defined by the SDP through an a=rtpmap attribute, or can be from the list of RFC-defined codecs. Examples are PCMU, opus, or telephone-event. Codecs stripped using this option are only removed from the outgoing rewritten SDP and don't affect the list of codecs that was offered by the source SDP. See the ignore option above for a similar mechanism that affects the offer codecs.

  It is possible to specify codec format parameters alongside with the codec name in the same format as they're written in SDP for codecs that support them, for example opus/48000 to specify Opus with 48 kHz sampling rate and one channel (mono), or opus/48000/2 for stereo Opus. If any format parameters are specified, the codec will only be stripped if all of the format parameters match, and other instances of the same codec with different format parameters will be left untouched.

  As a special keyword, all can be used to remove all codecs, except the ones that should explicitly offered (see below). Note that it is an error to strip all codecs and leave none that could be offered. In this case, the original list of codecs will be left unchanged.

  The keyword full can also be used, which behaves the same as all with the exception listed under transcode below.

• transcode

  Similar to offer but allows codecs to be added to the list of offered codecs even if they were not present in the original list of codecs. In this case, the transcoding engine will be engaged. Only codecs that are supported for both decoding and encoding can be added in this manner. This also has the side effect of automatically stripping all unsupported codecs from the list of offered codecs, as rtpengine must expect to receive or even send in any codec that is present in the list.

  Note that using this option does not necessarily always engage the transcoding engine. If all codecs given in the transcode list were present in the original list of offered codecs, then no transcoding will be done. Also note that if transcoding takes place, in-kernel forwarding is disabled for this media stream and all processing happens in userspace.

  If no codec format parameters are specified in this list (e.g. just opus instead of opus/48000/2), default values will be chosen for them.

  For codecs that support different bitrates, it can be specified by appending another slash followed by the bitrate in bits per second, e.g. opus/48000/2/32000. In this case, all format parameters (clock rate, channels) must also be specified.

  Additional options that can be appended to the codec string with additional slashes are ptime, the fmtp string, and additional codec-specific options. For example iLBC/8000/1///mode=30 to use mode=30 as fmtp string.

  For Opus, the string of codec-specific options is passed directly to ffmpeg, so all ffmpeg codec options can be set. Use space, colon, semicolon, or comma to separate individual options. For example to set the encoding complexity (also known as compression level by ffmpeg): opus/48000/2////compression_level=2

  If a literal = cannot be used due to parsing constraints (i.e. being wrongly interpreted as a key-value pair), it can be escaped by using two dashes instead, e.g. iLBC/8000/1///mode--30.

  As a special case, if the strip=all or mask=all option has been used and the transcode option is used on a codec that was originally present in the offer, then rtpengine will treat this codec the same as if it had been used with the offer option, i.e. it will simply restore it from the list of stripped codecs and won't actually engage transcoding for this codec. On the other hand, if a codec has been stripped explicitly by name using the strip or mask option and then used again with the transcode option, then the codec will not simply be restored from the list of stripped codecs, but instead a new transcoded instance of the codec will be inserted into the offer. (This special exception does not apply to mask=full or strip=full.)

  This option is only processed in offer messages and ignored otherwise.

Optionally included SDP attributes manipulations:

sdp-attr contains a dictionary controlling various aspects of attribute lines (or a= lines).

An intention of these option flags is to control session (global) and media level attributes, with help of which it's possible to do the following manipulations:

• addition
• removal
• substitution (replacement)

This does affect an outgoing SDP offer. So it's meant to manipulate body attributes, which rtpengine generates during the offer processing. In other words, it manipulates what has been already prepared by rtpengine on its own, taking into account received offer.

Furthermore, it's quite important to remember, that the changes, which have been applied to SDP body attributes, will be not taken into account by rtpengine itself, so these changes are rather formal (textual). This means, it's not the same, as if they would be originally given by the session originator.

That's why this kind of flags must be used with a full carefulness, because if not, this can potentially lead to the unexpected result.

Usage syntax:

	"sdp-attr" :
	{
		"<media-type>":
		{
			"<command>": ["<value>", "<value>"],
			"<command>": ["<value>", "<value>"]
		},
		"<media-type>":
		{
			"<command>": ["<value>", "<value>"],
			"<command>": ["<value>", "<value>"]
		}
	}

Description:

• <media-type>

  Defines a level of command application. One media type can be given only once per command. <media-type> can have one of the following values:

  • none or global

    Applies to the session level (global) attributes, but not to any of the media session specific attributes.

  • audio

    Applies to all currently present media sessions of audio type.

  • video

    Applies to all currently present media sessions of audio type.

• <command>

  The command to be applied to the targeted attribute line(s). Each command can be used multiple times within one media session/global scope.

  • add

    Adds a new a= line with a given value to the concerned attributes list. If the attribute with such value already exists within this scope of media session, then no duplication is to be added, therefore the older one remains untouched and nothing extra is being added.

    Can take multiple values (so multiple attributes can added per one command).

  • remove

    Removes a specified a= line from the concerned attributes list. If such line hasn't been found, then the attributes list remains untouched.

    The matching can be done using just the attribute name, or the attribute name plus a tag value, or the full attribute line (value and the following attribute parameters, if given). For example, the attribute a=foo:bar baz quuz would match any of foo, foo:bar, or foo:bar baz quuz`.

    Can take multiple values (so multiple attributes can removed per one command).

  • substitute

    Substitutes a specified a= line taken from the concerned media attributes list. If such line hasn't been found, then the attributes list remains untouched.

    The matching can be done using just the attribute name, or the attribute name plus a tag value, or the full attribute line (value and the following attribute parameters, if given). If the attribute is generated by rtpengine itself and a tag value is present, then the tag value must also be used in the match pattern. For example, the attribute a=foo:bar baz quuz would match any of foo, foo:bar, or foo:bar baz quuz, but the self-generated attribute a=fmtp:10 foobarcould only be substituted using eitherfmtp:10orfmtp:10 foobarbut not justfmtp`.

    Substitutes one attribute at a time, so one attribute into another attribute. Read more about that below in the <value> section.

• <value>

  The value has to not include the a= (lvalue) part. It contains only the value, that is given after the equal sign.

  No wild-cards or regular expressions are accepted.

  It's important to remember that some attributes are allowed to be present multiple times. Furthermore rtpengine does not expect given a= lines (to be substituted) to be unique within concerned media scope (global, audio or video).

  This leads to the next point — remove and substitute commands can affect just a single attribute, as well as multiple attributes, depending on the uniqueness of the value in the given command.

  User is supposed to provide full a= line value, so that it gives expected behavior.

  Important remark regarding substitute command. It takes only two values at a time, in other words it substitutes one attribute per command:

  • the first value, that matches the full value to be substituted; and

  • the second value, that is to be placed instead. Therefore, the only allowed syntax for it is (per command):

    "substitute": ["from-this-attribute", "to-that-attribute"]

  All other possible usages will be ignored and only first two values will be taken. However, multiple substitute commands can be given per time, see examples below.

Examples:

• Add a new (single) attribute line to the session (global) level:

    "sdp-attr" :
    {
    	"none" :
    	{
    		"add" : [ "sendrecv" ]
    	}
    }
  

• Add two new attribute lines to audio session and remove one for video session:

    "sdp-attr" :
    {
    	"audio" :
    	{
    		"add" : [ "ptime:20", "sendrecv" ]
    	},
    	"video":
    	{
    		"remove" : [ "rtpmap:101 telephone-event/8000" ]
    	}
    }
  

• Substitute two attributes of the global session and one for audio media section (pay attention, substitute uses lists, not dictionaries):

    "sdp-attr" :
    {
    	"none" :
    	{
    		"substitute": [[ "sendrecv" , "sendonly" ], [ "ptime:20" , "ptime:40" ]]
    	},
    	"audio" :
    	{
    		"substitute": [["fmtp:101 0-15" , "fmtp:126 0-16" ]]
    	},
    }
  

• As an alternative syntax these can be listed in the flags list. An example of such syntax:

• sdp-attr-remove-audio-ptime:20
• sdp-attr-substitude-none-sendrecv>sendonly.

In such usage equals sign (=) can be escaped as double dashes (--) and spaces can be escaped as double periods (..).

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" ],
"address family": "IP6", "received-from": [ "IP4", "10.65.31.43" ],
"ICE": "force", "transport protocol": "RTP/SAVPF", "media address": "2001:d8::6f24:65b",
"DTLS": "passive" }

A response message contains only 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=..." }

Optionally included SDP media manipulations:

sdp-media-remove contains a list pointing, which media types are to be removed from SDP.

This does affect an outgoing SDP offer. So it's meant to manipulate an SDP body, which rtpengine generates during the offer processing. The removed media type will be then not taken into consideration during further processing.

When this flag is added, rtpengine will not show concerned media type(s), hence media section(s) to the recipient's side. Therefore, later on the recipient side will provide only an answer for those media section(s) shown to it.

Upon processing such an answer coming back to the changed SDP offer, rtpengine will just add a zeroed media towards the originator's side in order to fulfill RFC requirements telling to use a zeroed media for those unaccepted media sections.

Usage syntax:

	"sdp-media-remove" : ["<media-type>", "<media-type>", ...]

Examples:

• Remove all occurences of the video media type:

    "sdp-media-remove" : ["video"]
  

IANA-registered media types are understood, or the special type other can be given to remove all media sections with types that are not understood.

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": ...).

• to-tag

  This flag controls whether the "To" tag's value is honoured or ignored when handling delete messages. Normally, the "To" tag's value is always included when present, but can be disregarded for the "delete" type of messages. So that, including the "To-tag" option flag in the "delete" message, forces to honour it and hence allows to be more selective about monologues within a dialog to be torn down.

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.

  • label

    The label assigned to this endpoint in the offer or answer 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 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.

If the optional recording-file key is set, then its value will be used as an output file. Note that the value must refer to a complete (absolute) path including file name, and a file name extension will not be added.

If the optional recording-dir key is set, then its value will be used as the directory path for the output file(s), overriding the output-dir config option of the recording daemon. The value should refer to an existing directory given as an absolute path. Setting this key does not affect the names of the files that will be created in the directory.

If the optional recording-pattern key is set, then its value will be used as the pattern to generate the output file name(s), overriding the output-pattern config option of the recording daemon. Note that no validity checking is performed on the given string, so make sure that the given pattern does not yield duplicate file names.

The option recording-file takes precedence over both recording-dir and recording-pattern if multiple options are set.

stop recording Message

The stop recording message must contain the key call-id as defined above. The reply dictionary contains no additional keys. See below under pause recording for another alternative usage for this message.

Disables call recording for the call. This can be sent during a call to immediately stop recording it.

pause recording Message

Identical to stop recording except that it instructs the recording daemon not to close the recording file, but instead leave it open so that recording can later be resumed via another start recording message.

Alternatively the stop recording message can be used if either the string pause is given in the flags list, or if the dictionary contains the key pause=yes.

block DTMF and unblock DTMF Messages

These message types must include the key call-id in the message. They enable and disable blocking of DTMF events (RFC 4733 type packets), respectively.

Packets can be blocked for an entire call if only the call-id key is present in the message, or can be blocked directionally for individual participants. Participants can be selected by their SIP tag if the from-tag key is included in the message, they can be selected by their SDP media address if the address key is included in the message, or they can be selected by the user-provided label if the label key is included in the message. For an address, it can be an IPv4 or IPv6 address, and any participant that is found to have a matching address advertised as their SDP media address will have their originating RTP packets blocked (or unblocked).

Unblocking packets for the entire call (i.e. only call-id is given) does not automatically unblock packets for participants which had their packets blocked directionally, unless the string all (equivalent to setting all=all) is included in the flags section of the message.

When DTMF blocking is enabled, DTMF event packets will not be forwarded to the receiving peer. If DTMF logging is enabled, DTMF events will still be logged to syslog while blocking is enabled. Blocking of DTMF events can be enabled and disabled at any time during call runtime.

block media and unblock media Messages

Analogous to block DTMF and unblock DTMF but blocks media packets instead of DTMF packets. DTMF packets can still pass through when media blocking is enabled. Media packets can be blocked for an entire call, or directionally for individual participants. See block DTMF above for details.

In addition to blocking media for just one call participant, it's possible to block media for just a single media flow. This is relevant to scenarios that involve forked media that were established with one or more subscribe request. To select just one media flow for media blocking, in addition to selecting a source call participant as above, a destination call participant must be specified using the to-tag or to-labelkey in the message.

Another possibility to block media for individual media flows is to use one of the special all= keywords instead of directly specifying a single to-tag or to-label. With all=offer-answer all media flows from the given from-tag that resulted from an offer/answer negotiation are affected. Respectively with all=except-offer-answer the opposite happens. With all=flows all currently established media flows are affected regardless or how they were created.

silence media and unsilence media Messages

Identical to block media and unblock media except that media packets are not simply blocked, but rather have their payload replaced with silence audio. This is only supported for certain trivial audio codecs (i.e. G.711, G.722).

start forwarding and stop forwarding Messages

These messages control the recording daemon's mechanism to forward PCM via TCP/TLS. Unlike the call recording mechanism, forwarding can be enabled for individual participants (directionally) only, therefore these messages can be used with the same options as the block and unblock messages above. The PCM forwarding mechanism is independent of the call recording mechanism, and so forwarding and recording can be started and stopped independently of each other.

play media Message

Only available if compiled with transcoding support. The message must contain the key call-id and one of the participant selection keys described under the block DTMF message (such as from-tag, address, or label). Alternatively, the all flag can be set to play the media to all involved call parties.

Starts playback of a provided media file to the selected call participant. The format of the media file can be anything that is supported by ffmpeg, for example a .wav or .mp3 file. It will automatically be resampled and transcoded to the appropriate sampling rate and codec. The selected participant's first listed (preferred) codec that is supported will be chosen for this purpose. Encoder parameters such as bit rate can be set via the codec-set list described above.

Media files can be provided through one of these keys:

• file

  Contains a string that points to a file on the local file system. File names can be relative to the daemon's working direction.

• blob

  Contains a binary blob (string) of the contents of a media file. Due to the limitations of the ng transport protocol, only very short files can be provided this way, and so this is primarily useful for testing and debugging.

• db-id

  Contains an integer. This requires the daemon to be configured for accessing a MySQL (or MariaDB) database via (at the minimum) the mysql-host and mysql-query config keys. The daemon will then retrieve the media file as a binary blob (not a file name!) from the database via the provided query.

• repeat-times

  Contains an integer. How many times to repeat playback of the media. Default is 1.

• repeat-duration

  Contains an integer. How much time in milliseconds is a playback of the media to be minimally iterated. E.g. if set to 10000ms and the playback's length is 1000ms, then this playback will be iterated 10 times due to limitation set to 10000ms. If used together with repeat-times then the following logic takes place: if repeat-duration hits the trigger earlier, then this playback will be stopped, otherwise if the repeat-duration is still positive, but the repeat-times counter went down to 1, then still the playback is to be stopped. By default is disabled.

• start-pos

  Contains an integer. The start frame position to begin the playback from.

In addition to the result key, the response dictionary may contain the key duration if the length of the media file could be determined. The duration is given as in integer representing milliseconds.

stop media Message

Stops the playback previously started by a play media message. Media playback stops automatically when the end of the media file is reached, so this message is only useful for prematurely stopping playback. The same participant selection keys as for the play media message can and must be used. Will return the last frame played in last-frame-pos key.

play DTMF Message

Instructs rtpengine to inject a DTMF tone or event into a running audio stream. A call participant must be selected in the same way as described under the play media message above (including the possibility of using the all flag). The selected call participant is the one generating the DTMF event, not the one receiving it.

The dictionary key code (or alternatively digit) must be present in the message, indicating the DTMF event to be generated. It can be either an integer with values 0-15, or a string containing a single character (0 - 9, *, #, A - D). Additional optional dictionary keys are: duration indicating the duration of the event in milliseconds (defaults to 250 ms, with a minimum of 100 and a maximum of 5000); volume indicating the volume in absolute decibels (defaults to -8 dB, with 0 being the maximum volume and positive integers being interpreted as negative); and pause indicating the pause in between consecutive DTMF events in milliseconds (defaults to 100 ms, with a minimum of 100 and a maximum of 5000).

This message can be used to implement application/dtmf-relay or application/dtmf payloads carried in SIP INFO messages. Multiple DTMF events can be queued up by issuing multiple consecutive play DTMF messages.

If the destination participant supports the telephone-event RTP payload type, then it will be used to send the DTMF event. Otherwise a PCM DTMF tone will be inserted into the audio stream. Audio samples received during a generated DTMF event will be suppressed.

The call must be marked for DTMF injection using the inject DTMF flag used in both offer and answer messages. Enabling this flag forces all audio to go through the transcoding engine, even if input and output codecs are the same (similar to DTMF transcoding, see above).

Music on hold functionality (MoH)

Only available if compiled with transcoding support.

This functionality is available only for the offer/answer model, hence no other scenarios like publish or subscription related are supported with it.

The concept — is that MoH capabilities get advertised always at the beginning of the call (original offer/answer exchange) and can be used later on to put the other side on hold.

MoH can be can be added for both sides: offerer and answerer. Hence, for the offerer one will advertisde MoH support when calling rtpengine_offer() (or rtpengine_manage()) and for the answerer side when calling rtpengine_answer() (or again rtpengine_manage()).

The one who advertises its MoH capabilities at the beginning of the call (first offer/answer exchange) can later trigger MoH using the sendonly state (or alternatively inactive) and then unhold the other side using sendrecv state (or alternatively no state advertised, which is equal to the sendrecv one).

MoH covers only audio type of media sessions, hence other media types, such as video, aren't accepted. It should also be taken into account that there is no specific selection of media sections to be held, thus — if one audio media puts the call on hold, the whole call is held.

During MoH being active, it's not possible to mix the MoH stream (sound file being played) with an audio stream coming from the one who puts on hold. In other words, all egress media stream coming from the MoH originator will be ignored until the other side gets unheld.

MoH also cannot be mixed with the play media functionality, the last one triggered will override previous one. Although, in this case the behavior of rtpengine can also get unexpected.

MoH must be unheld to stop the media being sent towards the other side (recipient). If MoH isn't unheld explicitly, then it will be stopped by the moh-max-duration config option.

To control a duration of MoH to be played, see moh-max-duration and moh-max-repeats configuration options. By default MoH playback is limited to 1800 seconds (half an hour).

MoH stream will also be dropped in case of full dialog termination.

The MoH SDP session can also be marked with a specific (custom) session level attribute. For that to enable see the moh-attr-name configuration option.

In order to protect against double MoH played, e.g. when inadvertently two rtpengine instances in the interaction chain both try to trigger MoH, use moh-prevent-double-hold configuration option (true/false). It works in combination with moh-attr-name config option, which must be defined in order to give a clue to rtpengine, what is the session level attribute to be used for a double hold detection.

List of parameters to be given when advertising MoH capabilities:

• a sound source : file, blob or db-id.

  At least this parameter must be given.

  file — is a full path to a file including name stored on the local file system.

  blob — is a blob data (binary data) directly given (in form of string) via option flags to rtpengine.

  db-id - an integer which points to a data stored in DB. At least mysql-host and mysql-query must be configured to let the system read blob data from the DB backend. For more information see play media functionality and mysql-* configuration options.

• mode : sendonly or sendrecv (by default always sendonly).

  Which state gets advertised to the other side, which is being put on hold. In some cases sendrecv can be useful for specific client implementations, which don't want to see sendonly state in coming SDP.

  mode also has an in-dialog-only value called reflect. This one isn't meant to advertise own capabilities and can only be used with in-dialog SDP offers, assuming that at the beginning of the call session MoH capabilities were advertised by the capable side.

  This mode is only supposed to be used in cases when:

  • in-dialog offerer doesn't have own MoH capabilities; and
  • still wants the recipient to hear the MoH music; hence
  • an offerer checks whether a recipient is capable of MoH and launches a player based on the given capabilities;
  • the rest functions the same as with usual MoH hold.

  mode=reflect contradicts with mode=sendrecv/mode=sendonly which in its turn serves another purpose.

• connection type : zero (for now only one type is available).

  If set, then connection information (c= field) in the according media session will be set to all zeroes (0.0.0.0). So called zeroed-hold. Can be useful for some older client implementations using outdated standards.

Usage syntax:

	"moh" :
	{
		"<source>": "<value>",
		"mode": "<value>",
		"connection": "<value>"
	}

Usage example:

	"moh" :
	{
		"db-id": "123456789",
		"mode": "sendrecv",
		"connection": "zero"
	}

Another usage example:

	"moh" :
	{
		"file": "/tmp/music-on-hold.wav"
	}

statistics Message

Returns a set of general statistics metrics with identical content and format as the list jsonstats CLI command. Sample return dictionary:

{
  "statistics": {
    "currentstatistics": {
      "sessionsown": 0,
      "sessionsforeign": 0,
      "sessionstotal": 0,
      "transcodedmedia": 0,
      "packetrate": 0,
      "byterate": 0,
      "errorrate": 0
    },
    "totalstatistics": {
      "uptime": "18",
      "managedsessions": 0,
      "rejectedsessions": 0,
      "timeoutsessions": 0,
      "silenttimeoutsessions": 0,
      "finaltimeoutsessions": 0,
      "offertimeoutsessions": 0,
      "regularterminatedsessions": 0,
      "forcedterminatedsessions": 0,
      "relayedpackets": 0,
      "relayedpacketerrors": 0,
      "zerowaystreams": 0,
      "onewaystreams": 0,
      "avgcallduration": "0.000000"
    },
    "intervalstatistics": {
      "totalcallsduration": "0.000000",
      "minmanagedsessions": 0,
      "maxmanagedsessions": 0,
      "minofferdelay": "0.000000",
      "maxofferdelay": "0.000000",
      "avgofferdelay": "0.000000",
      "minanswerdelay": "0.000000",
      "maxanswerdelay": "0.000000",
      "avganswerdelay": "0.000000",
      "mindeletedelay": "0.000000",
      "maxdeletedelay": "0.000000",
      "avgdeletedelay": "0.000000",
      "minofferrequestrate": 0,
      "maxofferrequestrate": 0,
      "avgofferrequestrate": 0,
      "minanswerrequestrate": 0,
      "maxanswerrequestrate": 0,
      "avganswerrequestrate": 0,
      "mindeleterequestrate": 0,
      "maxdeleterequestrate": 0,
      "avgdeleterequestrate": 0
    },
    "controlstatistics": {
      "proxies": [
	{
	  "proxy": "127.0.0.1",
	  "pingcount": 0,
	  "offercount": 0,
	  "answercount": 0,
	  "deletecount": 0,
	  "querycount": 0,
	  "listcount": 0,
	  "startreccount": 0,
	  "stopreccount": 0,
	  "startfwdcount": 0,
	  "stopfwdcount": 0,
	  "blkdtmfcount": 0,
	  "unblkdtmfcount": 0,
	  "blkmedia": 0,
	  "unblkmedia": 0,
	  "playmedia": 0,
	  "stopmedia": 0,
	  "playdtmf": 0,
	  "statistics": 0,
	  "errorcount": 0
	}
      ],
      "totalpingcount": 0,
      "totaloffercount": 0,
      "totalanswercount": 0,
      "totaldeletecount": 0,
      "totalquerycount": 0,
      "totallistcount": 0,
      "totalstartreccount": 0,
      "totalstopreccount": 0,
      "totalstartfwdcount": 0,
      "totalstopfwdcount": 0,
      "totalblkdtmfcount": 0,
      "totalunblkdtmfcount": 0,
      "totalblkmedia": 0,
      "totalunblkmedia": 0,
      "totalplaymedia": 0,
      "totalstopmedia": 0,
      "totalplaydtmf": 0,
      "totalstatistics": 0,
      "totalerrorcount": 0
    }
  },
  "result": "ok"
}

publish Message

Similar to an offer message except that it is used outside of an offer/answer scenario. The media described by the SDP is published to rtpengine directly, and other peer can then subscribe to the published media to receive a copy.

The message must include the key sdp which should describe sendonly media; and the key call-id and from-tag to identify the publisher. Most other keys and options supported by offer are also supported for publish.

The reply message will contain an answer SDP in sdp, but unlike with offer this is not a rewritten version of the received SDP, but rather a recvonly answer SDP generated by rtpengine locally. Only one codec for each media section will be listed, and by default this will be the first supported codec from the published media. This can be influenced with the codec options described above, in particular the accept option.

The list of codecs given in the accept option is treated as a list of codec preferences, with the first codec listed being the most preferred codec to be accepted, and so on. It is allowable to list codecs that are not supported for transcoding. If no codecs from the accept list are present in the offer, then the first codec that is supported for transcoding is selected. If no such codec is present, then the offer is rejected. The special string any can be given in the accept list to influence this behaviour: If any is listed, then the first codec from the offer is accepted even if it's not supported for transcoding.

subscribe request Message

This message is used to request subscription (i.e. receiving a copy of the media) to one or multiple existing call participants, which must have been created either through the offer/answer mechanism, or through the publish mechanism.

A single call participant can be selected in the same way as described under block DTMF. Multiple call participants can be selected either by using the all keyword, in which case all call participants that were created through the offer/answer mechanism will be selected, or by providing a list of tags (from-tags) in the from-tags list.

This message then creates a new call participant, which corresponds to the subscription. This new call participant will be identified by a newly generated unique tag, or by the tag given in the to-tag key. If a label is to be set for the newly created subscription, it can be set through set-label.

The reply message will contain a sendonly offer SDP in sdp which by default will mirror the SDP of the call participant being subscribed to. If multiple call participants are subscribed to at the same time, then this SDP will contain multiple media sections, combined out of the media sections of all selected call participants. This offer SDP can be manipulated with the same flags as used in an offer message, including the option to manipulate the codecs. The reply message will also contain the from-tags (corresponding to the call participants being subscribed to) and the to-tag (corresponding to the subscription, either generated or taken from the received message).

If a subscribe request is made for an existing to-tag then all existing subscriptions for that to-tag are deleted before the new subscriptions are created.

subscribe answer Message

This message is expected to be received after responding to a subscribe request message. The message should contain the same to-tag as the reply to the subscribe request as well as the answer SDP in sdp.

By default, the answer SDP must accept all codecs that were presented in the offer SDP (given in the reply to subscribe request). If not all codecs were accepted, then the subscribe answer will be rejected. This behaviour can be changed by including the allow transcoding flag in the message. If this flag is present, then the answer SDP will be accepted as long as at least one valid codec is present, and the media will be transcoded as required. This also holds true if some codecs were added for transcoding in the subscribe request message, which means that allow transcoding must always be included in subscribe answer if any transcoding is to be allowed.

The reply message will simply indicate success or failure. If successful, media forwarding will start to the endpoint given in the answer SDP.

unsubscribe Message

This message is a counterpart to subsscribe answer to stop an established subscription. The subscription to be stopped is identified by the to-tag.

connect Message

This message makes it posible to directly connect the media of two call parties without the need for a full offer/answer exchange. The required keys are call-id to identify the call, and from-tag and to-tag` to identify the two call parties to connect. The media will be connected in the same way as it would through an offer/answer exchange. Transcoding will automaticaly be engaged if needed.

It's also possible to connect two call parties from two different calls (different call IDs). To do so, the second call ID must be given as to-call-id, with the given to-tag then being one of the call parties from that second call. Internally, both calls will be merged into a single call object, with both call IDs then corresponding to the same call. This will be visible in certain statistics (e.g. two call IDs appearing in the list, but only one call being counted).