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.

Credentials

title

<title>Account Title</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

<username>johndoe</username>
Default:No default value
Description:SIP user name

password

<password>secretPhrase</password>
Default:No default value
Description:SIP password

userDisplayName

<userDisplayName>John Doe</userDisplayName>
Default:empty string
Description:SIP display user name, comes quoted in front of SIP uri.

Also displayed in WebRTC for the logged-in user (e.g. in the user menu or in the stream history).

authUsername

<authUsername>johndoe</authUsername>
Default:empty string
Description:Username used in SIP authorization headers. If left empty, the username is used.

host

<host>sip.example.com[:port]</host>
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 <proxy> is not specified. If no port is specified, 5060 is the default for non-encrypted transports, 5061 for tls transports.

transport

<transport>udp</transport>
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

<userCallerId>John Doe</userCallerId>
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

<expires>600</expires>
Default:600
Valid values:integer in range from 0 to 2 32-1.
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.

allowMessage

<allowMessage>1</allowMessage>
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

<sipisHost>sipis.example.com</sipisHost>
Default:sipis.acrobits.cz
Valid values:domain name or IP address
Description:Specifies the SIPIS server which the app should connect to.

sipisRegServer

<sipisRegServer>sipis.example.com</sipisRegServer>
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

<sipisHostPrefixLength>0</sipisHostPrefixLength>
Default:1
Valid values:integer values larger than 0
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

<proxy>proxy.example.com[:port]</proxy>
Default:empty string
Description:Address of SIP proxy server to connect to. If no port is specified, 5060 is used.

outboundProxy_enabled

<outboundProxy_enabled>0</outboundProxy_enabled>
Default:empty string
Description:when set to 1, all SIP traffic is routed through outboundProxy_host using outboundProxy_transport

outboundProxy_host

<outboundProxy_host>proxy.example.com[:port]</outboundProxy_host>
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

<outboundProxy_transport>udp</outboundProxy_transport>
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 <proxy> is empty, address = <host>, otherwise address = <proxy>

  • 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

<voiceMailNumber>*69</voiceMailNumber>
Default:empty string
Description:The number which is dialed when the “voice mail” button is pressed.

subscribeForVoicemail

<subscribeForVoicemail>1</subscribeForVoicemail>
Default:0
Valid values:1 or 0
Description:Set to send SUBSCRIBE for voice mail

Audio Codecs

codecOrder

<codecOrder>103,9,0,8,18,102,3</codecOrder>
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

<codecOrder3G>18,103,102,3,9,0,8</codecOrder3G>
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.

codecOrderWebrtc

<codecOrderWebrtc>103,9,0,8</codecOrderWebrtc>
Default:103,9,0,8
Valid values:comma separated list of codec numbers.
Description:The order of voice codecs, as it will be offered in SDP for calls in WebRTC apps. WebRTC only supports G711, G722 and Opus voice codecs.

honorTheirCodecListWiFi

<honorTheirCodecListWiFi>0</honorTheirCodecListWiFi>
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

<honorTheirCodecList3G>0</honorTheirCodecList3G>
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

<ptime>20</ptime>
Default:20
Description:packet time in milliseconds, to be used with WiFi codec set.

ptime3G

<ptime3G>30</ptime3G>
Default:30
Description:packet time in milliseconds, to be used with 3G codec set.

forcePtime

<forcePtime>0</forcePtime>
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

<forcePtime3G>0</forcePtime3G>
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:

<opusOptions3G.class>nb</opusOptions3G.class>

or

<opusOptions.class></opusOptions.class>
<opusOptions.bandwidth>fb</opusOptions.bandwidth>
<opusOptions.complexity>5</opusOptions.complexity>
<opusOptions.bitrate>300000</opusOptions.bitrate>
<opusOptions.fec>1</opusOptions.fec>
<opusOptions.dtx>0</opusOptions.dtx>
<opusOptions.vbr>0</opusOptions.vbr>

opusOptions.class

<opusOptions.class>wb</opusOptions.class>
<opusOptions3G.class>nb</opusOptions3G.class>
Default:

wb
Valid values:

nb, wb, fb or empty string
Description:

If set to nb, it’s equivalent to setting:

If set to wb, it’s equivalent to setting:

If set to fb, it’s equivalent to setting:

If set to empty string, you need to set other options yourself

opusOptions.bandwidth

<opusOptions.bandwidth>wb</opusOptions.bandwidth>
<opusOptions3G.bandwidth>nb</opusOptions3G.bandwidth>
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

<opusOptions.complexity></opusOptions.complexity>
<opusOptions3G.complexity>5</opusOptions3G.complexity>
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

<opusOptions.bitrate>20500</opusOptions.bitrate>
<opusOptions3G.bitrate>15500</opusOptions3G.bitrate>
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

<opusOptions.fec>1</opusOptions.fec>
<opusOptions3G.fec>1</opusOptions3G.fec>
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

<opusOptions.dtx>1</opusOptions.dtx>
<opusOptions3G.dtx>1</opusOptions3G.dtx>
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

<opusOptions.vbr>1</opusOptions.vbr>
<opusOptions3G.vbr>1</opusOptions3G.vbr>
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

<opusOptions.expectedPacketLoss>5</opusOptions.expectedPacketLoss>
<opusOptions3G.expectedPacketLoss>5</opusOptions3G.expectedPacketLoss>
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

videoCodecOrder

<videoCodecOrder>108,34,99</videoCodecOrder>
Default:

108,34,99
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
34  : H263
99  : H264

videoCodecOrder3G

<videoCodecOrder3G>108,34,99</videoCodecOrder3G>
Default:108,34,99
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.

videoCodecOrderWebrtc

<videoCodecOrderWebrtc>108,109,99</videoCodecOrderWebrtc>
Default:109,108,99
Valid values:comma separated list of codec numbers
Description:The order of video codecs, as it will be offered in SDP for calls made in WebRTC app. WebRTC doesn’t support H263 codec, but supports VP9 (use payload number 109).

videoMinKbpsWifi and videoMaxKbpsWifi

<videoMinKbpsWifi>64</videoMinKbpsWifi>
<videoMaxKbpsWifi>384</videoMaxKbpsWifi>
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

<videoMinKbps3G>64</videoMinKbps3G>
<videoMaxKbps3G>384</videoMaxKbps3G>
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

<videoDimsWifi>qcif</videoDimsWifi>
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

H263 codec only supports qcif and cif resolutions.

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

<videoDims3G>qcif</videoDims3G>
Default:qcif
Description:Specifies video resolution to be sent over cellular networks
Valid values:same as for videoDimsWifi

videoFPSWifi

<videoFPSWifi>15</videoFPSWifi>
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. Choosing high frame rate means more CPU power is needed to encode the video and less bits can be allocated to a single frame within the bandwidth constraints. When a bit rate is too high, some frames may be dropped if the CPU can’t keep up with encoding.

iOS camera has maximum fram rate of 30 fps. The pre-defined fps ranges from 10 for 3GS device to 20 for iPhone 4S.

videoFPS3G

<videoFPS3G>15</videoFPS3G>
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

<autoSendVideoWifi>1</autoSendVideoWifi>
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

<autoSendVideo3G>1</autoSendVideo3G>
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

<autoReceiveVideoWifi>1</autoReceiveVideoWifi>
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

<autoReceiveVideo3G>1</autoReceiveVideo3G>
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

<dtmfOrder>rfc2833,info,audio</dtmfOrder>
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. WebRTC does not support audio method.

Note

To disable a method, simply don’t put it on the list. The list may have only a single item.

dtmfAll

<dtmfAll>0</dtmfAll>
Default:0
Valid values:0 and 1
Description:When set, all supported DTMF modes will be sent together. Not recommended. WebRTC does not support this key.

rfc2833NegotiateOnly8kHzClockRate

<rfc2833NegotiateOnly8kHzClockRate>1</rfc2833NegotiateOnly8kHzClockRate>
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

<rfc2833EnforceDurationIn8KHzTimestampUnits>0</rfc2833EnforceDurationIn8KHzTimestampUnits>
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

<STUN>stun.example.com:[port]</STUN>
Default:empty string
Description:STUN server to use. When empty, libsoftphone will make a DNS SRV query for _stun._udp.sipserver and uses the returned address. If no port is specified, 3478 is used.

STUNUsername

<STUNUsername>user</STUNUsername>
Default:empty string
Description:Username for STUN/TURN server. If filled in, we’ll consider the STUN server TURN-capable.

STUNPassword

<STUNPassword>password</STUNPassword>
Default:empty
Description:Password for STUN/TURN server.

natTraversal

<natTraversal>auto</natTraversal>
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

<ignoreSymmetricNat>1</ignoreSymmetricNat>
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

<iceDefaultCandidateOrder>relay,srflx,host</iceDefaultCandidateOrder>
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

<contactIP>internal</contactIP>
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

<forcedContact>192.168.100.100:7777</forcedContact>
Default:empty string
Description:This IP address will be used in Contact and Via headers in case contactIP is set to “static”.

sendAudioBack

<sendAudioBack>0</sendAudioBack>
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

<keepAlive>1</keepAlive>
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

<keepAlivePeriod>30</keepAlivePeriod>
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

<rtpPortRangeStart>10000<rtpPortRangeStart>
<rtpPortRangeEnd>65530<rtpPortRangeEnd>
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

<listeningPort>5060</listeningPort>
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

<maxTimeToWaitForWorkingNetworkInMilliseconds>15000</maxTimeToWaitForWorkingNetworkInMilliseconds>
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

<headers>
    <header method="INVITE" name="X-CustomHeader" value="value"/>
    <header method="*" name="X-OtherHeader" value="12345"/>
…
</headers>

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

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:

<doNotDisturb gmtOffsetInMinutes="60">
  <dndEntry>
    <from>08:00</from>
    <to>10:00</to>
    <weekdays>31</weekdays>
    <contacts>
      <contactInfo
        name="+1 (222) 333-4455"
        phone="+12223334455"
        identifier=""
        source=""/>
    </contacts>
  </dndEntry>
  <dndEntry>
    <from>00:00</from>
    <to>23:59</to>
    <weekdays>96</weekdays>
    <contacts/>
    <enabled>1</enabled>
    <comment>Do not disturb on weekends</comment>
  </dndEntry>
</doNotDisturb>

The DND XML is wrapped in an <doNotDisturb> node. The optional attribute gmtOffsetInMinutes defines the timezone for the time values in DND entries. Inside this XML node, there is a list of <dndEntry> child nodes.

<from> and <to> 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 <from>00:00</from> and <to>23:59</to>.

<weekdays> 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.

<contacts> is a list of <contactInfo> child nodes. In case there are no <contactInfo> 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 <contactInfo> child nodes.

In case the <contactInfo> element has non-empty fields <identifier> and <source>, 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 <phone> 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 <name> is only a display name shown in DND rule editor GUI.

<enabled> is an optional tag. If not present or if its value is <enabled>1</enabled>, the rule is enabled, otherwise it is disabled.

Optional tag <comment> 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

<rewriting>
   <rule>
     <conditions>
       <condition type="doesntStartWith" param="+"/>
       <condition type="longerThan" param="8"/>
     </conditions>
     <actions>
       <action type="prepend" param="+420"/>
     </actions>
   </rule>
   <rule>
     … another rule …
   </rule>
 …
</rewriting>
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
  2. doesntStartWith
    matches numbers that doesn’t start with the parameter
  3. equals
    dialed number is exactly the same as the parameter
  4. lengthEquals
    the length of the dialed number equals the parameter
  5. shorterThan
    the length of the dialed number is shorter then the parameter
  6. longerThan
    the length of the dialed number is longer then the parameter
  7. networkType
    parameter is one of wifi, cellular, any, none. Matches if the current network is the one specified by parameter
  8. 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
  2. prepend
    prepends the parameter to the dialed number
  3. append
    appends the parameter to the dialed number
  4. continue
    no parameter, try following rules as well
  5. recordCall
    turns on call recording for this call
  6. 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)
  7. confirm
    unused
  8. drop
    unused
  9. callThrough
    forces a call-through dial action for the matching number
  10. 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 Dial Actions for more details.
  11. setHeader
    sets SIP header for outgoing SIP call. The param contains the header to be set, for example: <action type="setHeader" param="X-Condition-Matched: true"/>.
  12. rejectCall
    used in Incoming Call Rewriter. Rejects the call (486 Busy Here).
  13. forwardCall
    used in Incoming Call Rewriter. Forwards the call to the number specified in param attribute.
  14. answerImmediately
    used in Incoming Call Rewriter. Answers the incoming call immediately, without any user interaction.

Incoming Calls [*]

icm_auto

Deprecated since version C4AEB: builds after Dec 2016.

See icm instead.

<icm_auto>1</icm_auto>

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 since version C4AEB: builds after Dec 2016.

See icm instead.

<incomingDisabled>1</incomingDisabled>

Valid Values: 0,1

Default Value: N/A

pushMethod

Deprecated since version C4AEB: (builds after Dec 2016).

See icm instead.

<pushMethod>tunnel</pushMethod>

Valid Values: off, tunnel

Enables Push Notifications for incoming calls.

Default: N/A

bgrEnabled

Deprecated since version C4AEB: builds after Dec 2016.

See icm instead.

<bgrEnabled>1</bgrEnabled>

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 since version C4AEB: builds after Dec 2016.

See icm instead.

<forceRegistration>1</forceRegistration>

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:

<icm_auto>0</icm_auto>
<pushMethod>tunnel</pushMethod>

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.

icm

New in version C4AEB: builds after Dec 2016.

<icm>auto</icm>

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

New in version C4AEB: builds after Dec 2016.

<requiresRegistrationForOutgoingCalls>1</requiresRegistrationForOutgoingCalls>

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

pushBlockReg

<pushBlockReg>1</pushBlockReg>

Default value: 0

Setting to 1 blocks registration on the device until any Pushed incoming call has ended.

remoteContact

<remoteContact>pai,from,rpid,ppi</remoteContact>

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, <empty>

Defalut: N/A

Example: <sdesIncoming>enabled</sdesIncoming>

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

<extProvUrl>https://my.server.com/prov.php?u=%account[username]%&amp;p=%account[password]%</extProvUrl>

Default value: empty.

Reprovisioning URL, the app will ask this URL for complete account details. The service is expected to return <account> with all the required properties.

extProvInterval

<extProvInterval>3600</extProvInterval>

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 Balance Checker for detailed description.

Properties:

genericBalanceCheckUrl
genericBalanceCheckPostData
genericBalanceCheckContentType
balanceCheckIntervalInSeconds
balanceCheckDelayInSeconds

Web Callback

See Web Callback for detailed description.

wcb_enabled

<wcb_enabled>1</wcb_enabled>

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

<cth_enabled>1</cth_enabled>

Default value: 0

This node is used to enable/disable callthrough service.

cth_dialString

<cth_dialString>1</cth_dialString>

Default value: 0

This dial string to dial via GSM for callthrough service. The dial string may use placeholder replacement described in Global parameters and Account parameters scopes.

cth_ws_enabled

<cth_ws_enabled>1</cth_ws_enabled>

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 Web Call-Through page.

Properties:

cth_ws_url
cth_ws_method
cth_ws_contentType
cth_ws_postData
cth_ws_customHeaders

Call Forwarding

forwardingEnabled

<forwardingEnabled>1</forwardingEnabled>

Default value: 0

When set to 1, any incoming call will be automatically forwarded to the number defined in forwardingNumber field. For WebRTC, this key enables forwarding UI - button in the incall modal window and automatic forwarding section in the settings.

forwardingNumber

<forwardingNumber>500</forwardingNumber>

Default value: empty.

All incoming calls are redirected to the number you specify. WebRTC ignores this key.

Miscellaneous

vpnStartupUrl

<vpnStartupUrl>https://my.server.com/dummy</vpnStartupUrl>

Default value: empty

If non empty, the web service is called before every registration attempt. Sometimes it’s required to bring VPN up.

customActionUrl

<customActionUrl>https://example.com/action?uri=%uri%&amp;user=%account[username]%</customActionUrl>

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 Dial Actions for more details and also Introduction 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( identifier )

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

<tryingTone>periodic(sine(50ms,500,440Hz),silence(40ms))</tryingTone>

Played when outgoing call is being establishes, before we get Ringing notification from the remote side.

ringingTone

<ringingTone>periodic(sine(2000ms,2500,440Hz),silence(4000ms))</ringingTone>

Played when outgoing call is ringing at the remote side.

endCallTone

<endCallTone>periodic(sine(300ms,1000,400Hz),silence(400ms),sine(300ms,1000,400Hz),silence(400ms),sine(300ms,1000,400Hz),silence(700000ms))</endCallTone>

Played to notify local user that the remote party hung up the call.

busyTone

<busyTone>periodic(sine(500ms,2000,400Hz),silence(500ms))</busyTone>

Played when we receive “Busy” answer on outgoing call.

callWaitingTone

<callWaitingTone>periodic(silence(1000ms),sine(300ms,7500,440Hz),silence(9000ms))</callWaitingTone>

Played in case there is an established call and another incoming call is waiting to be answered.

zrtpHandshakeTone

<zrtpHandshakeTone>periodic(sine(50ms,500,400Hz,450Hz),silence(80ms))</zrtpHandshakeTone>

Played during ZRTP handshake.

Auto Answer

autoAnswerEnabled

<autoAnswerEnabled>1</autoAnswerEnabled>
Default:0
Description:Determines if the incoming call should be answered automatically. If enabled (value 1), the call is answered according to autoAnswerTimeoutInSeconds, or in case the incoming SIP call contains header Call-Info: with answer-after=xx, the timeout will be taken from this SIP header.

autoAnswerTimeoutInSeconds

<autoAnswerTimeoutInSeconds>5</autoAnswerTimeoutInSeconds>
Default:0
Description:If autoAnswerEnabled is 1, the amount of time to wait before answering the call automatically. A value of 0 means the call is immediately answered. If the incoming SIP call contains header Call-Info: with answer-after=xx, the timeout value will be taken from this SIP header instead.

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

<blf>
    <uri>1001</uri>
    <uri>1002</uri>
    <uri>5000</uri>
</blf>
Default:<empty>
Description:blf node contains list of uris/extensions. The app will subscribe to the listed uris for dialog-info events.

Note

For External Provisioning, if the incoming XML contains the <blf> node with some <uri> entries, the list from incoming XML replaces the existing list, discarding any previous values (if the priorities allow that, see Mergeable XML. It is also possible to provision using “union” operation, as in: <blf priority=”40” operation=”subnode-union”>....</blf> in which case the lists are merged together.

WebRTC Options

disableScreenSharing

<disableScreenSharing>1</disableScreenSharing>
Default:0
Description:Hides screensharing button in the desktop app.

genericSmsFetchInterval

<genericSmsFetchInterval>8000</genericSmsFetchInterval>
Default:8000
Description:The time in milliseconds for polling interval of fetching web messages.

genericSmsFetchInBackgroundInterval

<genericSmsFetchInBackgroundInterval>20000</genericSmsFetchInBackgroundInterval>
Default:20000
Description:The time in milliseconds for polling interval of fetching web messages when the WebRTC app is in the background. When WebRTC runs in the browser, this will get slowed down by Chrome background tab throttling.

dscpPacketMarking

<dscpPacketMarking>1</dscpPacketMarking>
Default:0
Description:When enabled, DSCP field in the IP packets sent by WebRTC will have value 46 (aka Expedited Forwarding).

Warning

This works only on Linux & MacOS. See Chromium bug report for status of development for Windows.

disableImForWebrtc

<disableImForWebrtc>0</disableImForWebrtc>
Default:0
Description:Disables sending of instant messages for WebRTC.

Account Example

<account>
    <username>xxx</username>
    <password>secret</password>
    <host>1.2.3.4</host>
    <codecOrder>9,0,8,102,3</codecOrder>
    <codecOrder3G>102,3,9,0,8</codecOrder3G>
</account>