.. _account-xml: =========== Account XML =========== SIP Accounts are specified in an XML format, where SIP credentials and various other options are specified. Below is the description of the recognized XML nodes and their possible values. .. contents:: :local: Credentials ----------- ``title`` ========= .. code-block:: xml Account Title :Default: empty :Description: The title of the account. It is shown at various places in the GUI and it must have a non-empty value. It is normally populated from Cloud Softphone portal or filled-in by user in the edit account forms. In case you set the value via provisioning, make sure it's not empty, otherwise the account validation will fail and the account won't be saved. ``username`` ============ .. code-block:: xml johndoe :Default: No default value :Description: SIP user name ``password`` ============ .. code-block:: xml secretPhrase :Default: No default value :Description: SIP password ``cloud_username`` and ``cloud_password`` ========================================= .. code-block:: xml johndoe@company.org secretPhrase2 :Default: No default value :Description: Many providers prefer to use different credentials to log in to the app than for the SIP credentials. If you return values for these, they can be used for authorizing external provisioning requests or other web service request as described here :ref: `client-api-intro`. ``userDisplayName`` =================== .. code-block:: xml John Doe :Default: empty string :Description: SIP display user name, comes quoted in front of SIP uri. ``authUsername`` ================ .. code-block:: xml johndoe :Default: empty string :Description: Username used in SIP authorization headers. If left empty, the username is used. ``host`` ======== .. code-block:: xml sip.example.com[:port] :Default: No default value. :Description: SIP domain, used in SIP URIs as ``username@host``. The SIP domain is also used as a SIP server to connect to, in case ```` is not specified. If no port is specified, 5060 is the default for non-encrypted transports, 5061 for tls transports. ``transport`` ============= .. code-block:: xml udp :Default: ``udp`` :Valid values: ``udp``, ``tcp``, ``tls`` and ``tls+sip:`` :Description: The transport protocol to use when communicating with SIP server. Transport ``tls`` uses sips: URIs in headers and the traffic is protected by TLS. ``tls+sip:`` is also TLS-protected, but uses sip: URIs. ``userCallerId`` ================ .. code-block:: xml John Doe :Default: empty string :Description: If set, this value will be sent in the ``From:`` field in outgoing ``INVITE`` requests. In case it's left empty, username will be used. ``expires`` =========== .. code-block:: xml 600 :Default: 600 :Valid values: integer larger or equal to 30 :Description: value for the Expires: header in SIP REGISTER packet. .. note:: Servers may reject registration with too low value in Expires: header and negotiate a higher value. .. warning:: If you're using Acrobits Cloud solution for incoming calls, the Acrobits' Sipis servers may reject/ban instances that require an expiration interval shorter than ``300`` seconds. Ask an Acrobits representative about the option of hosting your own Sipis servers in case your deployment requires a more frequent registration refresh. ``subscriptionExpires`` =========== .. code-block:: xml 300 :Default: 300 :Valid values: integer larger or equal to 30 :Description: value for the Expires: header in SUBSCRIBE packets. ``allowMessage`` ================ .. code-block:: xml 1 :Default: ``0`` :Valid values: ``0`` and ``1`` :Description: When set, ``MESSAGE`` will be added to the ``Allow:`` list in ``SIP REGISTER`` packets. SIP/SIMPLE messaging won't work without this set. SIPIS ----- ``sipisHost`` ============= .. code-block:: xml sipis.example.com :Default: ``sipis.acrobits.cz`` :Valid values: domain name or IP address :Description: Specifies the SIPIS server which the app should connect to. ``sipisRegServer`` ================== .. code-block:: xml sipis.example.com :Default: ``reg.acrobits.cz`` :Valid values: domain name or IP address :Description: Specifies the server which the app will use to transfer the account credentials to SIPIS. Typically it will be the same address as in ``sipisHost`` ``sipisHostPrefixLength`` ========================= .. code-block:: xml 0 :Default: ``1`` :Valid values: nonnegative integer values :Description: Used for load-balancing of SIPIS servers. In case it's greater than zero, the actual ``sipisHost`` value is constructed by prefixing ``s-X.`` to address in ``sipisHost``. The X is taken from account selector, which is SHA1(username:password:host). The value in sipisHostPrefixLength specifies how many letters from the hex-digit representation of selector to use. For example, with sipisHostPrefixLength=1, the app will use servers ``s-0.sipis.example.com`` - ``s-F.sipis.example.com``. You will need to set up the DNS and deploy multiple SIPIS servers to make load balancing work. .. note:: Note that the default value is ``1``. If you want to use a single SIPIS server (which is OK for up to 20,000 users on any reasonable machine), you will need to set this explicitly to ``0``. Proxies ------- ``proxy`` ========= .. code-block:: xml proxy.example.com[:port] :Default: empty string :Description: Address of SIP proxy server to connect to. If no port is specified, 5060 is used. ``outboundProxy_enabled`` ========================= .. code-block:: xml 0 :Default: empty string :Description: when set to 1, all SIP traffic is routed through ``outboundProxy_host`` using ``outboundProxy_transport`` ``outboundProxy_host`` ====================== .. code-block:: xml proxy.example.com[:port] :Default: empty string :Description: Address through which all SIP traffic is routed. If no port is specified, 5060 is used. outboundProxy_enabled needs to be set to ``1`` for this option to take effect. ``outboundProxy_transport`` =========================== .. code-block:: xml udp :Default: udp :Description: Transport to use for outboud proxy. All values described in ``transport`` field may be used. outboundProxy_enabled needs to be set to ``1`` for this option to take effect. .. note:: The final address of the SIP server is determined as follows: - **if** outbound proxy is enabled and its host is filled it, all traffic is routed through it - **if** ```` is empty, ``address = ``, otherwise ``address = `` - **if** address is not numeric (a.b.c.d) and port is not specified in address, do a SRV DNS resolution for ``_sip._udp.address`` (or ``_sip._tcp.address``, depending on the transport used) - **if** SRV record exists, ``address = DNSResultAddress:DNSResultPort`` - **else** use address Voicemail --------- ``voiceMailNumber`` =================== .. code-block:: xml *69 :Default: empty string :Description: The number which is dialed when the "voice mail" button is pressed. ``subscribeForVoicemail`` ========================= .. code-block:: xml 1 :Default: ``0`` :Valid values: ``1`` or ``0`` :Description: Set to send ``SUBSCRIBE`` for voice mail ``pushVoicemail`` ========================= .. code-block:: xml 1 :Default: ``0`` :Valid values: ``1`` or ``0`` :Description: Set to 1 if you want new voicemails reported to SIPIS to trigger a push notification to the device. Audio Codecs ------------ ``codecOrder`` ============== .. code-block:: xml 103,9,0,8,18,102,3 :Default: 103,9,0,8,18,102,3 :Valid values: comma separated list of codec numbers. :Description: The order of codecs, as it will be offered in SDP for calls made over Wifi network. Supported codecs are: :: 0 : G711 μ-law 3 : GSM 8 : G711 a-law 9 : G722 wideband codec 102 : iLBC 18 : G729 103 : Opus ``codecOrder3G`` ================ .. code-block:: xml 18,103,102,3,9,0,8 :Default: 18,103,102,3,9,0,8 :Valid values: comma separated list of codec numbers. :Description: The order of codecs, as it will be offered in SDP for calls made over cellular network. See the `codecOrder`_ field above for the list of valid codec payloads. ``honorTheirCodecListWiFi`` =========================== .. code-block:: xml 0 :Default: ``1`` :Valid values: ``0`` or ``1`` :Description: Forces the application to use the first codec offered by the remote side for calls over Wifi ``honorTheirCodecList3G`` ========================= .. code-block:: xml 0 :Default: ``0`` :Valid values: ``0`` or ``1`` :Description: Forces the application to use the first codec offered by the remote side for calls over 3G .. warning:: It's not recommended to change `honorTheirCodecList3G`_ and `honorTheirCodecListWiFi`_ defaults. .. note:: Consider an incoming INVITE with codecs "0,18" in SDP. SIP account is configured with `codecOrder`_ set to "0,18" and `codecOrder3G`_ "18,0". If we're on cellular network, it makes sense to ignore the remote's preference of codec "0" (μ-law), because we know our bandwidth is limited. This is why `honorTheirCodecList3G`_ if ``0`` by default - with this setting, we pick the first codec on our list, which is also supported by the remote side ("18"-G.729) On the other hand, when on wifi connection, we assume that we have enough bandwidth to honor the remote's preference, therefore the default value is ``1``. ``ptime`` ========= .. code-block:: xml 20 :Default: 20 :Description: packet time in milliseconds, to be used with WiFi codec set. ``ptime3G`` =========== .. code-block:: xml 30 :Default: 30 :Description: packet time in milliseconds, to be used with 3G codec set. ``forcePtime`` ============== .. code-block:: xml 0 :Default: ``0`` :Valid values: ``0`` or ``1`` :Description: If set to ``1``, the `ptime`_ parameter will be used even if the remote side requests a different value. ``forcePtime3G`` ================ .. code-block:: xml 0 :Default: ``0`` :Valid values: ``0`` or ``1`` :Description: If set to ``1``, the `ptime3G`_ parameter will be used even if the remote side requests a different value. Opus codec options ------------------ Opus codec parameters may be fine-tuned via several keys. The keys start either with `opusOptions.` or `opusOptions3G.` for WiFi or cellular calling respectively. You can either just set the `opusOptions.class`_ which can be ``nb`` for narrowband, ``wb`` for wideband, or ``fb`` for fullband. Alternatively you can finetune all the opus parameters. In this case the `opusOptions.class`_ parameter need to be set to the empty string. Then you can set `opusOptions.bandwidth`_, `opusOptions.complexity`_, `opusOptions.bitrate`_, `opusOptions.fec`_, `opusOptions.vbr`_, `opusOptions.dtx`_, and `opusOptions.expectedPacketLoss`_. Example: .. code-block:: xml nb or fb 5 300000 1 0 0 ``opusOptions.class`` ================================================ .. code-block:: xml wb nb :Default: ``wb`` :Valid values: ``nb``, ``wb``, ``fb`` or empty string :Description: If set to ``nb``, it's equivalent to setting: - `opusOptions.bandwidth`_ to ``nb`` - `opusOptions.complexity`_ to empty string - `opusOptions.bitrate`_ to ``15500`` - `opusOptions.fec`_ to ``1`` - `opusOptions.dtx`_ to ``1`` - `opusOptions.vbr`_ to ``1`` - `opusOptions.expectedPacketLoss`_ to ``5`` If set to ``wb``, it's equivalent to setting: - `opusOptions.bandwidth`_ to ``wb`` - `opusOptions.complexity`_ to empty string - `opusOptions.bitrate`_ to ``24000`` - `opusOptions.fec`_ to ``1`` - `opusOptions.dtx`_ to ``1`` - `opusOptions.vbr`_ to ``1`` - `opusOptions.expectedPacketLoss`_ to ``5`` If set to ``fb``, it's equivalent to setting: - `opusOptions.bandwidth`_ to ``fb`` - `opusOptions.complexity`_ to empty string - `opusOptions.bitrate`_ to ``64000`` - `opusOptions.fec`_ to ``1`` - `opusOptions.dtx`_ to ``1`` - `opusOptions.vbr`_ to ``1`` - `opusOptions.expectedPacketLoss`_ to ``5`` If set to empty string, you need to set other options yourself ``opusOptions.bandwidth`` ========================= .. code-block:: xml wb nb :Default: ``wb`` :Valid values: ``nb``, ``wb``, ``fb`` :Description: Limits the band width of the codec. - ``nb`` for narrowband i.e. 8kHz - ``wb`` for wideband i.e. 16kHz - ``fb`` for fullband i.e. 48kHz - any other value is treated as ``wb`` The hertz value will be signalled in SDP as ``maxplaybackrate``. Encoder will be set to the minimum of this setting and the value received in the SDP. ``opusOptions.complexity`` ========================== .. code-block:: xml 5 :Default: empty string :Valid values: 0 to 10 or empty string :Description: Configures the encoder's computational complexity. The supported range is 0-10 inclusive with 10 representing the highest complexity. When not specified, use Opus default. ``opusOptions.bitrate`` ======================= .. code-block:: xml 20500 15500 :Default: empty string :Valid values: integers between ``6000`` and ``510000``, empty string value, ``-1000``, or ``-1`` :Description: Desired encoder bitrate in bits per second. Values between ``6000`` and ``510000`` are meaningful. Empty string or ``-1000`` means automatic bitrate. Value ``-1`` means maximum bitrate. This value will be signalled in SDP. If received in SDP as ``maxaveragebitrate``, the value from SDP will be used for encoder. ``opusOptions.fec`` =================== .. code-block:: xml 1 1 :Default: ``0`` :Valid values: ``1`` or ``0`` :Description: Forward error correction. When enabled, encoder transmits redundant information which makes it possible to reconstruct lost packets on the received side. If enabled ``useinbandfec=1`` is signalled in SDP. The FEC for the encoder is enabled only if ``useinbandfec=1`` is received in SDP. Otherwise it is disabled. ``opusOptions.dtx`` =================== .. code-block:: xml 1 1 :Default: ``0`` :Valid values: ``1`` or ``0`` :Description: Enables discontinuous transmission (DTX). If enabled ``usedtx=1`` is signalled in SDP. The DTX for the encoder is enabled only if ``usedtx=1`` is received in SDP. Otherwise it is disabled. ``opusOptions.vbr`` =================== .. code-block:: xml 1 1 :Default: ``1`` :Valid values: ``1`` or ``0`` :Description: Enables variable bit rate. Note it's not recommended to use VBR for secure calls. If disabled ``cbr=1`` is signalled in SDP. The VBR for the encoder is disabled if ``cbr=1`` is received in SDP. Otherwise it is enabled. ``opusOptions.expectedPacketLoss`` ================================== .. code-block:: xml 5 5 :Default: ``0`` :Valid values: 0 - 100 :Description: Specifies the expected percentage of packet loss. Higher values will allocate more of the bit rate for redundant data to be used to reconstruct lost packets, which may degrade quality in the loss-less case (but will provide better quality in situations with lost packets) Video Codecs ------------ ``allowVideo`` =================== .. code-block:: xml 1 :Default: ``1`` :Valid values: ``1`` or ``0`` :Description: Value ``1`` enables video calls in the application, ``0`` disables video calls. ``videoCodecOrder`` =================== .. code-block:: xml 108,99 :Default: ``108,99`` :Valid values: comma separated list of codec numbers :Description: The order of video codecs, as it will be offered in SDP for calls made with Wifi codec set. Supported codecs are:: 108 : VP8 99 : H264 ``videoCodecOrder3G`` ===================== .. code-block:: xml 108,99 :Default: ``108,99`` :Valid values: comma separated list of codec numbers :Description: The order of video codecs, as it will be offered in SDP for calls made with 3G codec set. ``videoMinKbpsWifi`` and ``videoMaxKbpsWifi`` ============================================= .. code-block:: xml 64 384 :Default: 64 - 384 :Description: The values are in kilobits per second. Bandwidth range for video in case it's being sent over wifi. The encoder is initially configured to produce bitrate in the middle of the given range and adjusts the bitrate adaptively towards either end of the range depending on the network conditions. ``videoMinKbps3G`` and ``videoMaxKbps3G`` ========================================= .. code-block:: xml 64 384 :Default: 64 - 192 :Description: The values are in kilobits per second. Bandwidth range for video in case it's being sent over cellular network. See `videoMinKbpsWifi and videoMaxKbpsWifi`_ for more details. ``videoDimsWifi`` ================= .. code-block:: xml qcif :Default: qcif :Description: Specifies video resolution to be sent over wifi :Valid values: one of the resolution names, specified in table below: +-------------------+-----------------------+ | resolution name | frame dimensions | +===================+=======================+ | qcif | 176 × 144 | +-------------------+-----------------------+ | cif | 352 × 288 | +-------------------+-----------------------+ | vga | 640 x 480 | +-------------------+-----------------------+ | cif4 | 704 × 576 | +-------------------+-----------------------+ | cif16 | 1408 × 1152 | +-------------------+-----------------------+ | 720p | 1280 × 720 | +-------------------+-----------------------+ | 1080p | 1920 x 1080 | +-------------------+-----------------------+ .. note:: H264 and VP8 support all resolutions and in case the device changes orientation from landscape to portrait or vice-versa, the resolution is automatically transposed. ``videoDims3G`` =============== .. code-block:: xml qcif :Default: qcif :Description: Specifies video resolution to be sent over cellular networks :Valid values: same as for `videoDimsWifi`_ ``videoFPSWifi`` ================ .. code-block:: xml 15 :Default: empty string :Description: Frame rate in frames-per-second which the video encoder should use to encode video streams transferred over Wifi. .. note:: When left empty, the library chooses the pre-defined value for the current device, which will be 20 FPS on all modern devices. ``videoFPS3G`` ============== .. code-block:: xml 15 :Default: empty string :Description: Frame rate in frames-per-second which the video encoder should use to encode video streams transferred over cellular networks. ``autoSendVideoWifi`` ===================== .. code-block:: xml 1 :Default: empty string :Valid values: ``0``, ``1`` or empty string :Description: when set to empty string, the outgoing video stream is added to the call based on global app settings. Specifying ``0`` or ``1`` will override these settings for this particular account. When set to ``1``, video stream will be added into SDP when making general call (dialAction: "autoCall") over wifi network or when answering incoming call without explicitly specifying media type (for example, by clicking on notification about incoming call) .. note:: You can always explicitly request a video call by using "videoCall" dialAction, or voice-only call by "voiceCall" dialAction. Also note that even if the call is started without video, it can be added later. ``autoSendVideo3G`` =================== .. code-block:: xml 1 :Default: empty string :Valid values: ``0``, ``1`` or empty string :Description: Automatically add outgoing video stream into SDP when making call over cellular network. See `autoSendVideoWifi`_ for more details. ``autoReceiveVideoWifi`` ======================== .. code-block:: xml 1 :Default: empty string :Valid values: ``0``, ``1`` or empty string :Description: Automatically allow incoming video stream when for calls over wifi. See `autoSendVideoWifi`_ for more details. ``autoReceiveVideo3G`` ====================== .. code-block:: xml 1 :Default: empty string :Valid values: ``0``, ``1`` or empty string :Description: Automatically allow incoming video stream when for calls over cellular networks. See `autoSendVideoWifi`_ for more details. DTMF ---- ``dtmfOrder`` ============= .. code-block:: xml rfc2833,info,audio :Default: ``rfc2833,info,audio`` :Valid values: comma separated list of: rfc2833, info and audio. :Description: The methods to use when generating DTMF events. The methods are tried in the order specified here. RFC 2833 method is only used when the remote side supports telephone-event RTP payload. .. note:: To disable a method, simply don't put it on the list. The list may have only a single item. ``dtmfAll`` =========== .. code-block:: xml 0 :Default: ``0`` :Valid values: ``0`` and ``1`` :Description: When set, all supported DTMF modes will be sent together. Not recommended. ``rfc2833NegotiateOnly8kHzClockRate`` ===================================== .. code-block:: xml 1 :Default: ``1`` :Valid values: ``0`` and ``1`` :Description: In theory, the RFC 2833 DTMF RTP packets should use the same clock rate as the selected audio codec: "DTMF digits and named telephone events are carried as part of the audio stream, and MUST use the same sequence number and time-stamp base as the regular audio channel to simplify the generation of audio waveforms at a gateway." -- RFC 2833 This means that if an application offers audio codecs with distinct clock rates (e.g. PCMU/8000 and opus/48000/2), it should also offer *multiple telephone-event payload types, each with a corresponding distinct clock rate* (e.g. telephone-event/8000 and telephone-event/48000). And depending on the final outcome of the audio codec negotiation, the application should then use proper payload number, corresponding to either telephone-event/8000 or telephone-event/48000 when generating RFC 2833 DTMF RTP packets. In practice, however, many systems cannot deal with anything else than "telephone-event/8000" regardless of the selected audio codec and its clock rate. This option therefore allows switching between RFC 2833 conforming mode (value ``0``), in which the application offers multiple telephone-event payload types, and legacy mode (value ``1``) in which only "telephone-event/8000" is offered and recognized. ``rfc2833EnforceDurationIn8KHzTimestampUnits`` ============================================== .. code-block:: xml 0 :Default: ``0`` :Valid values: ``0`` and ``1`` :Description: In case the `rfc2833NegotiateOnly8kHzClockRate`_ option is enabled and the clock rate of the negotiated audio codec differs from 8 kHz, the app can populate the duration field of RFC 2833 DTMF RTP packets with either unmodified timestamp units corresponding to the audio codec clock rate (value ``0``), or it can adjust the timestamp units to correspond to 8 kHz clock rate (value ``1``). NAT Traversal ------------- ``STUN`` ======== .. code-block:: xml stun.example.com:[port] :Default: empty string :Description: STUN server to use. When empty, libsoftphone will use ``stun.acrobits.cz``. If no port is specified, 3478 is used. ``STUNUsername`` ================ .. code-block:: xml user :Default: empty string :Description: Username for STUN/TURN server. If filled in, we'll consider the STUN server TURN-capable. ``STUNPassword`` ================ .. code-block:: xml password :Default: empty :Description: Password for STUN/TURN server. ``natTraversal`` ================ .. code-block:: xml auto :Default: auto :Valid values: off, auto, stunOnly, turnAlways, ice :Description: Specifies the NAT traversal mode to use. This setting controls which addresses will be put into SDP. The options are described in detail below: :off: will always send local/private IP address detected from local network interface. No public address resolution is done. :auto: will do a STUN query for a public address. If symmetric NAT is not detected, the global address is used. In case of symmetric NAT, TURN server is used if STUNUsername is filled in, otherwise private/local address is used. :stun: will do a STUN query and always use the obtained public address (unless symmetric NAT is detected or if `ignoreSymmetricNat`_ is set to 1) :turnAlways: will always use TURN server :ice: will use the ICE process to choose the best address ``ignoreSymmetricNat`` ====================== .. code-block:: xml 1 :Default: ``0`` :Valid values: ``0`` and ``1`` :Description: if set, it will not check for symmetric NAT and will use the public IP if instructed so even when on symmetric NAT. Setting it to 1 is not recommended, since the IP:port detected will most likely not be accessible from outside, but sometimes it may be necessary to send the public IP even though the port is wrong. ``iceDefaultCandidateOrder`` ============================ .. code-block:: xml relay,srflx,host :Default: relay,srflx,host :Valid values: comma-separated list of "relay","srflx","host" :Description: You can prioritise one default ICE candidate over others. For example to prefer server reflexive address over relay address, set it to srflx,host,relay. To prefer host address, set it to host,srflx,relay. ``contactIP`` ============= .. code-block:: xml internal :Default: internal :Valid values: internal, external, static :Description: Specifies which address to use in Contact and Via SIP headers: :internal: will use the local/private in the Contact and Via SIP header. :external: will try to determine the public address using STUN ping, OPTIONS packet or STUN server and use it in Contact and Via header. :static: will use a static address in Contact and Via header, the address is specified in `forcedContact`_ node. ``forcedContact`` ================= .. code-block:: xml 192.168.100.100:7777 :Default: empty string :Description: This IP address will be used in Contact and Via headers in case `contactIP`_ is set to "static". ``sendAudioBack`` ================= .. code-block:: xml 0 :Defalut: ``1`` :Valid values: ``0`` and ``1`` :Description: If set to ``1``, we will always send the audio back to the address from which we receive the audio from the other side. ``keepAlive`` ============= .. code-block:: xml 1 :Default value: ``0`` :Valid values: ``0`` and ``1`` :Description: When set, libSoftphone periodically sends keep-alive packet to SIP server, to keep the connection active and NAT port mappings open. ``keepAlivePeriod`` =================== .. code-block:: xml 30 :Default value: 30 :Valid values: integer number. The value is in seconds, the minimum allowed value is 5 :Description: Specifies the amount of seconds between two keep-alive packets ``rtpPortRangeStart`` and ``rtpPortRangeEnd`` ============================================= .. code-block:: xml 10000 65530 :Default: 10000 and 65535. :Valid values: integer numbers between 1025 and 65535 (inclusive). There should be at least 4 ports between these values :Description: The port range to use for RTP sockets. ``listeningPort`` ================= .. code-block:: xml 5060 :Default: empty string :Description: The local port which the SIP stack will bind to: :Valid values: integer numbers between 1025 and 65535 (inclusive) | - If left empty, the SIP stack will choose a random port number and will bind to it. This number will remain the same for at least one day. If the client is restarted and re-registers repeatedly, it will remember its rinstance, callId and CSeq numbers, port will remain the same and therefore it should always occupy the same registration slot on the server. - If set to 0, a random port will be chosen every time the client registers. - If set to a specific numeric value, the SIP stack will always bind to that port. .. warning:: On iOS, setting `listeningPort`_ for TCP and TLS SIP transports has no effect. Specifying a port to bind to for TCP is not supported by iOS platform. ``maxTimeToWaitForWorkingNetworkInMilliseconds`` ================================================ .. code-block:: xml 15000 :Default: 15000 :Description: the time in milliseconds for which the client keeps calls in established state after the network is lost. When this time expires and there is still no network, the call is hung up. Custom SIP Headers ------------------ .. code-block:: xml
… Specify custom headers to be added into SIP packets. The headers should be stored in their own node named "headers" as sub-nodes with name "header". The attributes are: :method: SIP request method. The custom header will be added only to SIP packets of the specified method. Use ``method="*"`` to add the header to all SIP packets. :name: header name :value: header value .. _dnd: Do Not Disturb -------------- Do Not Disturb (DND), also called Zen Mode in Android Developers Documentation, mutes all events when the user does not want to be disturbed. DND rules can be provisioned using the XML format described below. Here is an example: .. code-block:: xml 08:00 10:00 31 00:00 23:59 96 1 Do not disturb on weekends The DND XML is wrapped in an ```` node. The optional attribute ``gmtOffsetInMinutes`` defines the timezone for the time values in DND entries. Inside this XML node, there is a list of ```` child nodes. ```` and ```` elements define the start and end time when the DND rule applies. The format is ``HH:MM`` and the hours are in 24h format. Both are required, if you want all-day rule, it should be ``00:00`` and ``23:59``. ```` is a bit mask which defines for which days in the week the rule applies. The values are as follows: ===== ======= =============== bit value day ===== ======= =============== 0 1 Monday 1 2 Tuesday 2 4 Wednesday 3 8 Thursday 4 16 Friday 5 32 Saturday 6 64 Sunday ===== ======= =============== If you are not familiar with bitwise arithmetics, all you need to do is to sum the values for days when you want the rule to apply. For example, weekends are Saturday and Sunday, i.e. 32+64 = 96. Workdays would be 1+2+4+8+16 = 31. Note that if you set this to 0, the rule will never be applied; to match any day, you need to set it to 127. ```` is a list of ```` child nodes. In case there are no ```` child nodes, the rule will match for any incoming number. In case the list is not empty, the rule applies only if the incoming party matches some of the ```` child nodes. In case the ```` element has non-empty fields ```` and ````, the rule will apply in case the incoming phone number matches address book contact from the given source and with the given identifier. For provisioning, you will typically not know these identifiers, so these will be usually left empty (you can also omit them). In such case, the app uses the value in field ```` to match against the incoming call and the rule applies only when these two are equivalent; it should be in E.164 format. The field ```` is only a display name shown in DND rule editor GUI. ```` is an optional tag. If not present or if its value is ``1``, the rule is enabled, otherwise it is disabled. Optional tag ```` can be used to add a comment to the rule. This is displayed in the GUI. In the example above, this DND setup will reject all calls coming from number +1 (222) 333-4455 between 8AM and 10AM (GMT+1) on workdays, and it will reject all calls from anyone on weekends. .. _number-rewriting: Number Rewriting ---------------- .. code-block:: xml … another rule … … :Default: none. Rules are examined in sequence until all conditions of a rule match the dialed number. Then all the actions are applied to the number. If neither of the actions is continue the transformation ends. However, if one of the actions of the rule is continue, another rule is examined and the conditions are now matched against the transformed number (not the originally dialed number). The example above means: Add +420 to numbers not starting with + that are longer than 8 digits. Condition nodes have two attributes: ``type`` (see below) and ``param`` - the parameter of the condition 1. ``startsWith`` matches numbers that start with the parameter #. ``doesntStartWith`` matches numbers that doesn't start with the parameter #. ``equals`` dialed number is exactly the same as the parameter #. ``lengthEquals`` the length of the dialed number equals the parameter #. ``shorterThan`` the length of the dialed number is shorter then the parameter #. ``longerThan`` the length of the dialed number is longer then the parameter #. ``networkType`` parameter is one of ``wifi``, ``cellular``, ``any``, ``none``. Matches if the current network is the one specified by parameter #. ``ssid`` matches if the current Wi-Fi SSID is the one specified in parameter Action nodes have two attributes: ``type`` (see below) and optional ``param`` - the parameter of the action. 1. ``replace`` applicable only to conditions ``startWith`` and ``equals``. Replaces the matched part with parameter #. ``prepend`` prepends the parameter to the dialed number #. ``append`` appends the parameter to the dialed number #. ``continue`` no parameter, try following rules as well #. ``recordCall`` turns on call recording for this call #. ``dialOut`` overrides dial-out account. In multi-account clients, this action makes it possible to route certain numbers through a specific account (for example, emergency numbers via cellular account) #. ``confirm`` unused #. ``drop`` unused #. ``callThrough`` forces a call-through dial action for the matching number #. ``overrideDialAction`` changes the dialAction to the one specified in param. This can be used to dial certain numbers using callThrough or using native gsm call, or show a list of options of what to do with the number. See :ref:`dialactions` for more details. #. ``setHeader`` sets SIP header for outgoing SIP call. The ``param`` contains the header to be set, for example: ````. #. ``rejectCall`` used in Incoming Call Rewriter. Rejects the call (486 Busy Here). #. ``forwardCall`` used in Incoming Call Rewriter. Forwards the call to the number specified in ``param`` attribute. #. ``answerImmediately`` used in Incoming Call Rewriter. Answers the incoming call immediately, without any user interaction. Incoming Calls [*]_ ------------------- ``icm_auto`` ============ .. deprecated:: C4AEB builds after Dec 2016. See `icm` instead. ``1`` Valid values: 0,1 Determines whether or not to use the global setting for incoming calls. Default value is 1 which means the global setting for incoming calls will be used. ``incomingDisabled`` ==================== .. deprecated:: C4AEB builds after Dec 2016. See `icm` instead. ``1`` Valid Values: 0,1 Default Value: N/A ``pushMethod`` ============== .. deprecated:: C4AEB (builds after Dec 2016). See `icm` instead. ``tunnel`` Valid Values: off, tunnel Enables Push Notifications for incoming calls. Default: N/A ``bgrEnabled`` ============== .. deprecated:: C4AEB builds after Dec 2016. See `icm` instead. ``1`` Valid Values: 0,1 Enables backgrounding for incoming calls (1 is enabled). Default Value: N/A Note: The transport protocol must be TCP or TLS for backgrounding to work. ``forceRegistration`` ===================== .. deprecated:: C4AEB builds after Dec 2016. See `icm` instead. ``1`` Valid Values: 0,1 Forces registration when incoming calls are off. Use 1 to force registration, only necessary if using incoming calls off with registration. Default Value: N/A .. note:: Only required properties are necessary. E.g., to set incoming calls to Push Notifications you would add:: 0 tunnel It is not necessary to add the other properties. .. [*] Not needed for external provisioning in the Cloud Softphone. Cloud Softphone providers should contact us through the `Cloud Softphone web portal`_ if they have any questions on how to set incoming calls. .. _`Cloud Softphone web portal`: https://providers.cloudsoftphone.com/ ``icm`` ======= .. versionadded:: C4AEB builds after Dec 2016. ``auto`` From Dec 2016, the incoming call mode configuration has been greatly simplified. The properties above have been deprecated (if you set them, the apps will still migrate them to the `icm` property though. Valid Values: auto, push, keepAwake, off Specifies the incoming call mode to be used for this account. If "auto" is set, the mode is taken from global app settings. Default Value: auto ``requiresRegistrationForOutgoingCalls`` ======================================== .. versionadded:: C4AEB builds after Dec 2016. ``1`` Valid Values: 0,1 When ``icm`` is set to ``off``, the app will not register for incoming calls. When it makes an outgoing call, some SIP backends will reject it because they only accept call attempts from registered endpoints. In case this parameter is set to 1, we will register the account when the outgoing call is made and keep it registered for the duration of the call. Default Value: 1 ``remoteContact`` ================= pai,from,rpid,ppi Default value: as above Specifies where in the incoming INVITE packet should we look for calling party identification. The value is a coma- separated list of identifiers described below. When incoming INVITE packet is received, we look for the remote party identification at places defined by the order here and use the first found value: +-----------+--------------------------------+ |identifier | SIP header | +===========+================================+ | pai | **P-Asserted-Identity** header | +-----------+--------------------------------+ | from | **From** header | +-----------+--------------------------------+ | rpid | **Remote-Party-ID** header | +-----------+--------------------------------+ | ppi | **P-Preferred-Identity** header| +-----------+--------------------------------+ Secure Calls ------------ Properties:: sdesIncoming sdesOutgoing zrtpIncoming zrtpOutgoing dtlsIncoming dtlsOutgoing Valid values: enabled, required, Defalut: N/A Example: ``enabled`` When set to ``enabled``, the app will use best-effort to establish the secure call using the selected technology if the other side supports it as well, but will fall back to unprotected call if it fails. In case of ``required``, the call will be dropped if security can't be established. If not set (left empty), secure calls using the corresponding technology won't be enabled even if the other side supports them - our side will behave as if the security protocol is not supported. External Provisioning --------------------- ``extProvUrl`` ============== ``https://my.server.com/prov.php?u=%account[username]%&p=%account[password]%`` Default value: empty. Reprovisioning URL, the app will ask this URL for complete account details. The service is expected to return ```` with all the required properties. ``extProvInterval`` =================== ``3600`` Default value: 0 How often (in seconds) will the app ask the reprovisioning service for account updates. If the value is 0, the periodic reprovisioning is disabled and reprovisioning URL is contacted only when saving the account on Edit Account screen. Balance Checking ---------------- See :ref:`balance_checker` for detailed description. Properties:: genericBalanceCheckUrl genericBalanceCheckPostData genericBalanceCheckContentType balanceCheckIntervalInSeconds balanceCheckDelayInSeconds Web Callback ------------ See :ref:`web_callback` for detailed description. ``wcb_enabled`` =============== ``1`` Default value: 0 This node is used to enable/disable web callback web service. Other properties:: wcb_url wcb_method wcb_contentType wcb_postData wcb_customHeaders Call Through ------------ ``cth_enabled`` =============== ``1`` Default value: 0 This node is used to enable/disable callthrough service. ``cth_dialString`` ================== ``+1555123456,%targetNumber%`` Default value: 0 This dial string to dial via GSM for callthrough service. The dial string may use placeholder replacement described in :ref:`global-params` and :ref:`account-params` scopes. ``cth_ws_enabled`` ================== ``1`` Default value: 0 This node is used to enable/disable callthrough service via a web service. The web service and its parameters are described in detail in :ref:`web_callthrough` page. Properties:: cth_ws_url cth_ws_method cth_ws_contentType cth_ws_postData cth_ws_customHeaders Call Forwarding --------------- ``forwardingEnabled`` ===================== ``1`` Default value: 0 When set to 1, any incoming call will be automatically forwarded to the number defined in `forwardingNumber` field. ``forwardingNumber`` ==================== ``500`` Default value: empty. All incoming calls are redirected to the number you specify. Call Waiting --------------- ``callWaiting`` ===================== ``1`` Default value: based on whether business features are enabled. With business features enabled, the default is set to 1. With business features disabled, the default is 0. Enables/disables call waiting. When set to 0, incoming calls will be rejected if user is already on an active call. Miscellaneous ------------- ``vpnStartupUrl`` ================= ``https://my.server.com/dummy`` Default value: empty If non empty, the web service is called before every registration attempt. Sometimes it's required to bring VPN up. ``customActionUrl`` =================== ``https://example.com/action?uri=%uri%&user=%account[username]%`` Default value: empty Defines the URL which is opened in external browser in case the ``openUrl`` Dial Action is invoked on a number or SIP uri. The placeholder ``%uri%`` is replaced by the actual dialed number. See :ref:`dialactions` for more details and also :ref:`client-api-intro` about web service parameter placeholders. Call Tones ---------- These are tones played locally to indicate call progress or status. The tones may be composed from elementary tones using periodic and compound operations. Durations are always in milliseconds, amplitudes can be either positive numbers, in which case they are used directly as amplitude of 16bit samples, or as negative values, interpreted as decibels (ITU-T recommendation is to use -24dBm0. Frequencies are in Hertz. Characters after digits are ignored, so it's a good idea to write the units after the number for clarity (see examples below). Elementary Tones ================ ``sine( duration, amplitude, freqs… )`` Creates a sine wave of given duration and amplitude. In case more than one frequency is specified, the tone will play the sine waves of given frequencies simultaneously. ``silence( duration )`` Plays a silence of given duration. ``wav( "filenameWithExtension" )`` Plays a WAV or CAF file. The files should be PCM or ADPCM (ima4) encoded, 8 or 16KHz. Stereo files are converted to mono. On iOS, it should specify filename with extension, the library will first search in ``NSApplicationSupportDirectory``, if not found, then in bundle ``resourcePath``. On Android, the parameter should be asset filename. Tone Operators ============== ``periodic( tones… )`` Plays the given tone segments in a loop. Tones may be any elementary tones, or compound tone. ``compound( tones… )`` Plays the given tones simultaneously. For simultaneous sine waves, a shortcut syntax of ``sine(…)`` with multiple frequencies is easier to write. ``tryingTone`` ``periodic(sine(50ms,500,440Hz),silence(40ms))`` Played when outgoing call is being establishes, before we get Ringing notification from the remote side. ``ringingTone`` ``periodic(sine(2000ms,2500,440Hz),silence(4000ms))`` Played when outgoing call is ringing at the remote side. For examples of common ringback tones refer to table below: +--------------------+----------------------------------------------------------------------------------------------------------+ | Country | Tone definition | +====================+==========================================================================================================+ | Canada | periodic(sine(2000ms,2500,440Hz,480Hz),silence(4000ms)) | +--------------------+----------------------------------------------------------------------------------------------------------+ | Italy | periodic(sine(1000ms,2500,425Hz),silence(4000ms)) | +--------------------+----------------------------------------------------------------------------------------------------------+ | Japan | periodic(sine(1000ms,2500,400Hz,416Hz),silence(2000ms)) | +--------------------+----------------------------------------------------------------------------------------------------------+ | UK | periodic(sine(400ms,2500,400Hz,450Hz),silence(200ms),sine(400ms,2500,400Hz,450Hz),silence(2000ms)) | +--------------------+----------------------------------------------------------------------------------------------------------+ | US | periodic(sine(2000ms,2500,440Hz,480Hz),silence(4000ms)) | +--------------------+----------------------------------------------------------------------------------------------------------+ ``endCallTone`` ``periodic(sine(300ms,1000,400Hz),silence(400ms),sine(300ms,1000,400Hz),silence(400ms),sine(300ms,1000,400Hz),silence(700000ms))`` Played to notify local user that the remote party hung up the call. ``busyTone`` ``periodic(sine(500ms,2000,400Hz),silence(500ms))`` Played when we receive “Busy” answer on outgoing call. For examples of common busy tones refer to table below: +--------------------+----------------------------------------------------------------------------------------------------------+ | Country | Tone definition | +====================+==========================================================================================================+ | UK | periodic(sine(375ms,2500,400Hz,450Hz),silence(375ms)) | +--------------------+----------------------------------------------------------------------------------------------------------+ | US | periodic(sine(500ms,2500,480Hz,620Hz),silence(500ms)) | +--------------------+----------------------------------------------------------------------------------------------------------+ ``callWaitingTone`` ``periodic(silence(1000ms),sine(300ms,7500,440Hz),silence(9000ms))`` Played in case there is an established call and another incoming call is waiting to be answered. ``zrtpHandshakeTone`` ``periodic(sine(50ms,500,400Hz,450Hz),silence(80ms))`` Played during ZRTP handshake. Auto Answer ----------- .. note:: If callkit is enabled on iOS we are unable to auto-answer the calls on lock screen or in background. ``appInitiatedAutoAnswerEnabled`` ================================= .. code-block:: xml 1 :Default: ``0`` :Description: Determines if the incoming call should be answered automatically. The application waits the amount of seconds configured via `autoAnswerTimeoutInSeconds`_ before answering the call automatically. ``autoAnswerTimeoutInSeconds`` ============================== .. code-block:: xml 5 :Default: ``0`` :Description: If `appInitiatedAutoAnswerEnabled`_ is ``1``, the amount of time to wait before answering the call automatically. A value of ``0`` means the call is immediately answered. ``sipInitiatedAutoAnswerEnabled`` ================================= .. code-block:: xml 1 :Default: ``0`` :Description: If enabled (value ``1``), the call is honoring the incoming SIP call header ``Call-Info:`` with ``answer-after=xx``, the timeout will be taken from this SIP header. Busy Lamp Field --------------- Busy Lamp Field allows to observe status of some phone extension on your device. For example, you can create a quick dial (favorite) entry and check that you are interested in Busy Lamp Field status. The app then shows an indicator whether this extension is on-hook or off-hook. On SIP level, the app has to SUBSCRIBE to on-hook and off-hook events from this extension (``dialog-info`` event package). The app therefore only receives info about subscribed extensions. The extensions to subscribe to are stored in account XML, in a format described below. ``blf`` ======= .. code-block:: xml 1001 1002 5000 :Default: :Description: ``blf`` node contains list of uris/extensions. The app will subscribe to the listed uris for dialog-info events. .. note:: For :ref:`external-provisioning`, if the incoming XML contains the node with some entries, the list from incoming XML replaces the existing list, discarding any previous values (if the priorities allow that, see :ref:`mergeable-xml`. It is also possible to provision using "union" operation, as in: `....` in which case the lists are merged together. CardDav ----------- ``cardDavTitle`` ========= .. code-block:: xml My CardDav :Default: No default value :Description: Title shown in the app for CardDav contacts, in Contacts tab. ``cardDavPrincipalUrl`` ============ .. code-block:: xml somecarddavurl.com :Default: No default value :Description: CardDAV principal URL. You can use placeholders like %account[username]% in the URL for any fields from Account XML. ``cardDavRefresh`` ============ .. code-block:: xml 180 :Default: 180 :Description: CardDAV refresh period in seconds. The app will check for any changes on the server periodically. ``cardDavUsername`` ============ .. code-block:: xml johndoe :Default: No default value :Description: Username for CardDav. If not sent, will default to use SIP username. ``cardDavPassword`` ============ .. code-block:: xml secrete :Default: No default value :Description: Password for CardDav. If not sent, will default to use SIP password. Push Options ---------------- .. warning:: The settings here are designed to fine-tune your account to avoid specific push call related issues with some SIP servers. Modifying them without encountering any problems may inadvertently cause new issues. It's strongly advised to only adjust these settings upon recommendation from Acrobits Support Team or if you have a thorough understanding of the implications for your SIP account. Proceed with caution. ``taglessSelector`` =========================== .. code-block:: xml 0 :Default: ``0`` :Description: By default, each registration on SIPIS for an account configured on multiple devices has its own selector. Setting to ``1`` will combine all instances into a single one with tokens for every device. This can help get calls to all devices provisioned with the same account even if you don’t support call forking/multiple registrations, but cannot solve all problem scenarios. e.g. there will still be multiple registrations if the app is open on any of the devices configured with the account. ``pushBlockReg`` =========================== .. code-block:: xml 0 :Default: ``0`` :Valid values: ``1`` or ``0`` :Description: By default, the app will register when it receives a pushed call. Setting to ``1`` will force the app to delay the registration until the pushed call is over. This can help with SIP servers that can’t handle new registrations (the app) while there is an active call on a different registration (SIPIS). ``pushNAT`` =========================== .. code-block:: xml 1 :Default: ``1`` :Valid values: ``1`` or ``0`` :Description: By default, SIPIS *pretends* to be behind NAT (even though it isn’t) because most SIP servers handle audio for pushed calls better in this case. Setting to ``0`` means that SIPIS will register using its public IP address. This can help with SIP servers that want the public IP in contact headers and rely on STUN. ``sipisMustUnregister`` =========================== .. code-block:: xml 0 :Default: ``0`` :Valid values: ``1`` or ``0`` :Description: By default, SIPIS does not unregister, it just lets registrations expire. Setting to ``1`` will force SIPIS to unregister before the app registers when it enters the foreground. This can help with SIP servers that need old registrations to unregister (SIPIS), before new ones are done (the app). Note: This is not normally needed even if the SIP server doesn’t support call forking or multiple registrations. As long as the SIP server sends the INVITE to the most recent registration (the normal behavior for servers that don’t support multiple registrations/call forking), push will work fine without setting this to ``1``. It will also add a slight delay to the registration from the app (since it must wait for SIPIS to unregister first). ``pushProxy`` =========================== .. code-block:: xml pushproxy.com:[port] :Description: Allows you to set a different proxy for SIPIS than the one the app uses in the foreground. Desktop Options ---------------- ``genericSmsFetchInterval`` =========================== .. code-block:: xml 8000 :Default: ``8000`` :Description: The time in milliseconds for polling interval of :doc:`fetching web messages`. ``genericSmsFetchInBackgroundInterval`` ======================================= .. code-block:: xml 20000 :Default: ``30000`` :Description: The time in milliseconds for polling interval of :doc:`fetching web messages` when the Desktop app is in the background. Video Conferencing ------------------ ``conferencingEnabled`` ======================== .. code-block:: xml 0 :Default: ``1`` :Description: When the value in the account.xml is set to "0" then Video Conferencing is explicitly disabled for the device. In case of more account.xmls at least one of these account.xmls must have value set to "0" to disable Video Conferencing for the device. Account Example --------------- :: xxx secret 1.2.3.4 9,0,8,102,3 102,3,9,0,8