Jump to: navigation, search

Configuring SIP Endpoint SDK for OS X

Important
The following features are not supported:
  • Video
  • SIP Cluster
  • SIP Proxy

Using the Default Configuration File

You can find the default configuration file in the following location:

<installation folder>/Bin/SipEndpoint.config

This file contains XML configuration details that affect how your SIP Endpoint SDK application behaves. The initial settings are the same as those specified for use with the QuickStart application included with your SIP Endpoint SDK release.

Configuration settings are separated into two containers: the Basic Container holds the connectivity details that are required to connect to your SIP Server, while the Genesys Container holds a variety of configuration settings.

Basic Container

The first Container ("Basic") holds the basic connectivity details that are required to connect to your SIP Server. This container has at least one connection (Connectivity) element with the following attributes:

<Connectivity user="DN" server="SERVER:PORT" protocol="TRANSPORT"/>


Make the following changes and save the updated configuration file before using the SIP Endpoint SDK:

  • user="DN" — Supply a valid DN for the user attribute.
  • server="SERVER:PORT" — Replace SERVER with the host name where your SIP Server is deployed, and PORT with the SIP port of the SIP Server host. (The default SIP port value is 5060.)
  • protocol="TRANSPORT" — Set the protocol attribute to reflect the protocol being used to communicate with SIP Server. Possible values are UDP, TCP, or TLS.

Genesys Container

The second Container ("Genesys") holds a number of configurable settings that are organized into domains and sections. These settings do not have to be changed, but can be customized to take full control over your SIP Endpoint SDK applications.

An overview of the settings in this container and the valid values for these settings is provided here:

Domain Section Setting
policy
endpoint
audio_qos
include_os_version_in_user_agent_header
include_sdk_version_in_user_agent_header
ip_versions
public_address
rtp_inactivity_timeout
rtp_port_min
rtp_port_max
signaling_qos
sip_port_min
sip_port_max
sip_transaction_timeout
vq_report_collector
vq_report_publish
vq_alarm_threshold
answer_sdp_priority
sip_port_binding
session
agc_mode
auto_answer
auto_answer_delay
dtmf_method
echo_control
noise_suppression
dtx_mode
reject_session_when_headset_na
sip_code_when_headset_na
vad_level
ringback_enabled
ringback_file
ringing_enabled
ringing_timeout
ringing_file
reject_session_when_busy
number_sessions_for_busy
sip_code_when_busy
rx_agc_mode
device
audio_in_device

For more information, see Audio Device Settings

audio_out_device
capture_device
ringer_device
headset_name
use_headset
codecs—See Working with Codec Priorities
proxies
proxy<n>
display_name
domain
password
reg_interval
reg_match_received_rport
reg_timeout
mailbox
password
server
timeout
transport
user
nat (sub-section of proxy<n>)
ice_enabled
stun_server
stun_server_port
turn_password
turn_relay_type
turn_server
turn_server_port
turn_user_name
system
diagnostics
enable_logging
log_file
log_filter
log_level
log_options_provider
logger_type
security
certificate
certificate-key
trusted-ca
sec-protocol
cipher-list
tls-crl
tls-target-name-check
use_srtp

policy Domain

endpoint Section


audio_qos

Valid Values: Number

The integer value representing the DSCP bits to set for RTP audio packets. The value should be 4 * Preferred DSCP value.


include_os_version_in_user_agent_header

Valid Values: 0 or 1
Default Value: 1

If set to 1, the user agent field includes the OS version the client is currently running on.


include_sdk_version_in_user_agent_header

Valid Values: 0 or 1
Default Value: 1

If set to 1, the user agent field includes the SDK version the client is currently running on.


ip_versions

Valid Values: IPv4, empty Default Value: IPv4.

A value of IPv4 or an empty value means that the application selects an available local IPv4 address.

NOTE: This parameter has no effect if the public_address option specifies an explicit IP address.


public_address

Valid Values: See description below.

Local IP address of the machine. This setting can be an explicit setting or a special value that the SDK uses to automatically obtain the public address.

Valid Values:
This setting may have one of the following explicit values:

  • An IP address. For example, 192.168.16.123 for IPv4.
  • A bare host name, for example, epsipmac2.

Default Value: Empty string which is fully equivalent to the $auto value.
This setting may have one of the following special values:

  • $auto—The SDK selects the first valid IP address on the first network adapter that is active (status=up) and has the default gateway configured. IP family preference is specified by the policy.endpoint.ip_versions setting.
  • $ipv4—Same behavior as the $auto setting but the SDK restricts the address to a particular IP family.
  • $host—The SDK retrieves the standard host name for the local computer using the gethostname system function.
  • An adapter name or part of an adapter name prefixed with $. For example, en0 or en1. The specified name must be different from the special values $auto, $ipv4, and $host.
  • $net:subnet - The SDK will select the IP address matching the given network ( from any local interface). The subnet is the full CIDR name as per RFC 4632. For example, $net:192.168.0.0/16.


If the value is specified as an explicit host name, the Contact header includes the host name for the recipient of SIP messages (SIP Server or SIP proxy) to resolve on their own. For all other cases, including $host, the resolved IP address is used for Contact. The value in SDP is always the IP address.

rtp_inactivity_timeout

Valid Values: Integer between 5 and 150
Default Value: 150
Suggested Value: 30

Timeout interval in seconds for RTP inactivity.

rtp_port_min

Valid Values: Number
Default Value: 8000

The integer value representing the minimum value for an RTP port range. Must be within the valid port range of 8000 to 65535. If the minimum value is not specified or set to an invalid value, the default value of 8000 is used for rtp_port_min. Setting the minimum to a value that is larger than the maximum is considered an error and will result in a failure to initialize the endpoint.


rtp_port_max

Valid Values: Number
Default Value: 9000

The integer value representing the maximum value for an RTP port range. Must be within the valid port range of 8000 to 65535. If the maximum value is not specified or set to an invalid value, the default maximum 9000 is used for rtp_port_max. Setting the maximum to a value that is less than the minimum is considered an error and results in a failure to initialize the endpoint.


signaling_qos

Valid Values: Number

The integer value representing the DSCP bits to set for SIP packets. Integer value should be configured to 4 * Preferred DSCP value.


sip_port_min

Valid Values: Number between 1 and 65535
Default Value: 5060

The integer value representing the minimum value for a SIP port range. Must be within the valid port range of 1 to 65535. If the minimum value is not specified or set to an invalid value, the default value of 5060 is used for sip_port_min. Setting the minimum to a value that is larger than the maximum is considered an error and will result in a failure to initialize the endpoint.


sip_port_max

Valid Values: Number between 1 and 65535
Default Value: 5080

The integer value representing the maximum value for a SIP port range. Must be within the valid port range of 1 to 65535. If the maximum value is not specified or set to an invalid value, the default value of 5080 is used for sip_port_max. Setting the maximum to a value that is less than the minimum is considered an error and will result in a failure to initialize the endpoint.


sip_transaction_timeout

Valid Values: Number between 1 and 32000
Default Value: 4000

SIP transaction timeout value in milliseconds. Valid values are 1 through 32000, with a default value of 4000. The recommended value is 4000.


vq_report_collector

See Producing RTCP Extended Reports

vq_report_publish

See Producing RTCP Extended Reports

vq_alarm_threshold

Valid Values: 0 or a number from 1.0 to 5.0

Default Value: 0

Specifies the MOS threshold for generating Voice Quality Alarms. A 0 value disables the alarms. The recommended threshold value is 3.5. Genesys recommends that you avoid using values above 4.2 as an MOS that high might not be obtainable with some codecs, even in perfect network conditions.


answer_sdp_priority

Valid Values: config or offer
Default value: config

  • config—The endpoint selects the first codec from the codec configuration listed in both the codec configuration and the SDP offer.
  • offer—the endpoint selects the first codec in the SDP offer listed in both the codec configuration and the SDP offer.

sip_port_binding

Valid Values: 0 or 1
Default Value: 0

  • 0—opens the SIP port to listen on any interface
  • 1—the SIP port binds to the interface specified by the public_address setting and listens only on this IP address.

session Section


agc_mode

Valid Values: 0 or 1


If set to 0, AGC (Automatic Gain Control) is disabled; if set to 1, it is enabled. Default: 1. Other values are reserved for future extensions. This configuration is applied at startup, after which time the agc_mode setting can be changed to 1 or 0 from the main sample application. NOTE: It is not possible to apply different AGC settings for different channels in multi-channel scenarios.


auto_answer

Valid Values: 0 or 1


If set to 1, all incoming calls should be answered automatically.


auto_answer_delay

Valid Values: Number in milliseconds

Default Value: 0

Time in milliseconds to wait before auto-answering (only applicable when auto_answer=1 and auto-answer is not blocked by missing headset).

dtmf_method

Valid Values: Rfc2833, Info, or InbandRtp

Method to send DTMF


echo_control

Valid Values: 0 or 1

If set to 1, echo control is enabled.


noise_suppression

Valid Values: 0 or 1

If set to 1, noise suppresion is enabled.


dtx_mode

Valid Values: 0 or 1

If set to 1, DTX is activated.


reject_session_when_headset_na

Valid Values:0 or 1

If set to 1, the SDK should reject the incoming session if a USB headset is not available.


sip_code_when_headset_na

Valid Values: Number

Defaul Value: 480
If a valid SIP error code is supplied, the SDK rejects the incoming session with the specified SIP error code if a USB headset is not available.


vad_level

Valid values: 0–3,from 0 (conventional VAD) to 3 (aggressive high)

Sets the degree of bandwidth reduction.

ringback_enabled

Valid Values: 0, 1, 2, 3

Default Value: 2

Specifies whether the ringback tone is enabled for outgoing calls.

  • 0 — The ringback is not played when the INVITE dialog is not yet established. In scenarios where ringback is provided by Media Server, the ringback tone would be still present.
  • 1 — The incoming media stream is played if provided by the Media gateway in a reliable provisional response with SDP.
  • 2 — A local file is used for the ringback.
  • 3 — The ringback is always played using either a local file or media provided by the gateway.


ringback_file

Valid Values: Empty or the path to the ringback sound file. The path can be a file in the current directory or the full path to the sound file.

Default Value: Empty

Specifies the audio file that is played when the ringing tone is enabled with the ringing_enabled option.

WebRTC does not support MP3 playback. The ringtone file for built-in ringback should be a RIFF (little-endian) WAVE file using one of the following formats:

  • kWavFormatPcm = 1, PCM, each sample of size bytes_per_sample
  • kWavFormatALaw = 6, 8-bit ITU-T G.711 A-law (8 KHz sampling rate)
  • kWavFormatMuLaw = 7, 8-bit ITU-T G.711 mu-law (8 KHz sampling rate)

Uncompressed PCM audio must be 16-bit mono or stereo with a sampling rate of 8, 16, or 32 KHz.

ringing_enabled

Valid values: 0, 1, 2, 3, 4, or 5.
Default Value: 3

  • 0—event Ringing disabled
  • 1—event Ringing enabled
  • 2—play ringtone internally (event Ringing disabled)
  • 3—play ringtone internally and enable event Ringing.
  • 4—play ringtone through separate "ringer" device
  • 5—play ringtone through "ringer" device and enable event Ringing


Suppressing the Ringtone

The ringtone is generated for all incoming calls to the Genesys SIP Endpoint SDK. To suppress the ringtone for third-party call control for the originating DN, configure the following SIP Server option:

  • make-call-alert-info=<urn:alert:service:3pcc@genesys>

or

  • make-call-alert-info=<file://null>;service=3pcc
Important
If at least one application based on SIP Endpoint SDK in the contact center is configured with the ringing_enabled option set to a non-zero value, the SIP Server make-call-alert-info option should be set to one of the specified values.

ringing_timeout

Valid Values: Empty, 0, or a positive number
Default Value: 0


Specifies the duration, in seconds, of the ringing tone. If set to 0 or if the value is empty, the ringing time is unlimited.


ringing_file

Valid Values: Empty string or string to the path to the ringing sound file. The path may be a file name in the current directory or the full path to the sound file.
Default Value: ringing.wav


Specifies the audio file that is played when the ringing tone is enabled with the ringing_enabled option.
Note that WebRTC does not support MP3 playback. The ringtone file for built-in ringing should be a RIFF (little-endian) WAVE file using one of the following formats:

kWavFormatPcm = 1, PCM, each sample of size bytes_per_sample
kWavFormatALaw = 6, 8-bit ITU-T G.711 A-law
kWavFormatMuLaw = 7, 8-bit ITU-T G.711 mu-law


Uncompressed PCM audio must be 16 bit mono or stereo and have a frequency of 8, 16, or 32 KHZ.


reject_session_when_busy

Valid Values: Empty, 0, 1

Default Value: 0

  • 0 or Empty—disable rejection of a session when busy
  • 1—enable rejection of a session when busy


number_sessions_for_busy

Valid Values: Positive integer

Default Value: 1

Sets the number of sessions before busy. Must be a positive integer.


sip_code_when_busy

Valid Values: Empty, 4xx, 5xx, 6xx

Default value: Empty

SIP error response code to use when busy. Can be set to any valid SIP error response code in the 4xx, 5xx, or 6xx range, for example, 486.

rx_agc_mode

Valid Values: 0, 1

Default value: 0

When set to 1, the SDK enables the receiving-side AGC allowing the volume of the received RTP stream to be adjusted automatically. When set to 0 (default), the feature is disabled.

device Section

audio_in_device

Valid Values: A regex that matches the ECMAScript standard.

Microphone device name. For more information, see Audio Device Settings


audio_out_device

Valid Values: A regex that matches the ECMAScript standard.

Speaker device name


capture_device

Valid Values: A regex that matches the ECMAScript standard.

Capture device name.


ringer_device

Valid Values: A regex that matches the ECMAScript standard.

Name of the device dedicated to playing ringing tone (used when ringing_enabled set to 4 or 5)

headset_name

Valid Values: A regex that matches the ECMAScript standard.

The name of the headset model.


use_headset

Valid values: 0 or 1

If set to 0, the audio devices specified in audio_in_device and audio_out_device are used by the SDK. If set to 1, the SDK uses a headset as the preferred audio input and output device and the audio devices specified in audio_in_device and audio_out_device are ignored.

codecs Section

See Working with Codec Priorities.

proxies Domain

proxy <n> Section


display_name

Valid Values: String

Proxy display name


domain

Valid Values: String containing any valid SIP domain
Default Value: Empty string

A SIP domain is an application layer configuration defining the management domain of a SIP proxy. The configured value should include hostport and may include uri-parameters as defined by RFC 3261. The scheme, userinfo, and transport URI parameters are included automatically.

If set to an empty string, SIP Endpoint SDK uses the parameters from the Connectivity section to construct the SIP domain value as it did in previous versions.


password

Valid Values: String

Proxy password. Password configured for DN object.


reg_interval

Valid Values: Integers greater than or equal to 0
Default Value: 0

The period, in seconds, after which the endpoint starts a new registration cycle when a SIP proxy is down. Valid values are integers greater than or equal to 0. If the setting is empty or negative, the default value is 0, which means no new registration cycle is allowed. If the setting is greater than 0, a new registration cycle is allowed and will start after the period specified by regInterval.


reg_match_received_rport

Valid Values: 0 or 1
Default Value: 0

This setting controls whether or not SIP Endpoint SDK should re-register itself when receiving a mismatched IP address in the received parameter of a REGISTER response. This helps resolve the case where SIP Endpoint SDK for OS X has multiple network interfaces and obtains the wrong local IP address. A value of 0 (default) disables this feature and a value of 1 enables re-registration.


reg_timeout

Valid Values: Number

The period, in seconds, after which registration should expire. A new REGISTER request will be sent before expiration. Valid values are integers greater than or equal to 0. If the setting is empty/null, then registration is disabled, putting the endpoint in standalone mode.

mailbox Section

Important
The mailbox section is a sub-section of the proxy<n> section.

password

Valid Values: String

Mailbox password


server

Valid Values: String

Proxy server address and port for this mailbox


timeout

Valid Values: Any positive integer
Default Value: 1800 seconds (30 minutes)

Subscription expiration timeout in seconds. If the setting is missing or set to 0, the SDK uses a default timeout of 1800 seconds (30 minutes).


transport

Valid Values: udp, tcp, or tls

Transport protocol to use when communicating with server


user

Valid Values: String

Mailbox ID for this mailbox

nat Sub-section

Important
nat is a sub-section of the proxy<n> section.


ice_enabled

Valid Values: Boolean

Enable or disable ICE.


stun_server

Valid Values: String

STUN server address. An empty or null value indicates this feature is not used.


stun_server_port

Valid Values: Valid port number

Default Value: 3478

STUN server port value.


turn_password

Valid Values: String

Password for TURN authentication.


turn_relay_type

Valid Values: 0, udp, 1, or tcp

Type of TURN relay.

  • 0 or udp for TURN over UDP.
  • 1 or tcp for TURN over TCP.


turn_server

Valid Values: String

TURN server address. An empty or null value indicates this feature is not used.


turn_server_port

Valid Values:Valid port number

Default Value: 3478

TURN server port value.


turn_user_name

Valid Values: String

User ID for TURN authorization

system Domain

diagnostics Section


enable_logging

Valid Values: 0 or 1

Valid values of 0 or 1 disable or enable logging.


log_file

Valid values: String containing full path or relative path to the log file.

Log file name, for example, SipEndpoint.log


log_filter

Valid Values: Empty, dtmf

Default Value: Empty

Specifies the list of log filters to be applied to hide sensitive data from the endpoint log. Currently the only supported filter is dtmf, which hides all occurrences of DTMF data from the log (by replacing entered digits with 'x').

log_level

Valid Values: Number, 04

Valid values correspond to log levels:

  • 0 = "Fatal"
  • 1 = "Error"
  • 2 = "Warning"
  • 3 = "Info"
  • 4 = "Debug".


log_options_provider

Valid Values: String

Valid values for webrtc are (warning, state, api, debug, info, error, critical). For example: gsip=2, webrtc=(error,critical)


logger_type

Valid Values: external

If set to external, an external logger is used.

security Section

Important
SIP Endpoint SDK no longer uses the tls_enabled setting.

certificate

Valid Values: String

Full path pointing to the certificate file used as a client-side certificate for outgoing TLS connections and as a server-side certificate for incoming TLS connections. For example, /users/Desktop/certificate/hostname_cer.pem This option replaces the cert_file option from previous versions. For backwards compatibility, the SDK accepts certificate or cert_file.

certificate-key

Valid Values: String

Full path to the endpoint private key file used for client-side authentication for outgoing TLS connections. For example, /users/Desktop/certificate/hostname_priv_key.pem

This option replaces the priv_key_file option from previous versions. For backwards compatibility, the SDK accepts certificate-key or priv_key_file.

trusted-ca

Valid Values: String

Full path to the Certificate Authority's list file. For example, /users/Desktop/certificate/ca/ca_cert.pem

This option replaces the ca_list_file option from previous versions. For backwards compatibility, the SDK accepts trusted-ca or ca_list_file.

sec-protocol

Valid Values: SSLv23, SSLv3, TLSv1, TLSv11, TLSv12

Default Value: SSLv23

Specifies the protocol used by the component to set up secure connections:

  • SSLv23 - The highest TLS protocol version supported by both sides of communication.
  • SSLv3 - SSL version 3.0.
  • TLSv1 - TLS version 1.0.
  • TLSv11 - TLS version 1.1.
  • TLSv12 - TLS version 1.2.

cipher-list

See the cipher-list setting defined in the Security Deployment Guide.

tls-crl

See the tls-crl setting defined in the Security Deployment Guide.

tls-target-name-check

Valid Values: no, host

Default Value: host

Specifies if the Common Name in the subject field and/or the Subject Alternate Names of the server’s certificate will be compared to the target host name (option value host). If they are not identical, the connection fails. If the option is set to no, a comparison is not made, and the connection is allowed.


use_srtp

Valid Values: optional, allowed, disabled, off, elective, both, enabled, force, mandatory

Indicates whether to use SRTP:

  • optional or allowed—do not send secure offers, but accept them
  • disabled or off—do not send secure offers and reject incoming secure offers
  • elective or both—send both secure and non-secure offers and accept either
  • enabled—send secure offers, accept both secure and non-secure offers
  • force or mandatory—send secure offers, reject incoming non-secure offers

Adding either ',UNENCRYPTED_SRTCP' (long form) or ',UEC' (short form) to any value (for example, "enabled,UEC""), would result in the UNENCRYPTED_SRTCP parameter being added to that offer. When this parameter is negotiated, RTCP packets are not encrypted, but are still authenticated.

Producing RTCP Extended Reports

You can use SIP Endpoint SDK to produce RTCP Extended Reports (RFC 3611) and publish them according to RFC 6035 at the end of each call, using a collector address of your choice.

Important
The publish message is sent to the specified collector address only if the vq_report_collector parameter is configured with the user@server:port;transport=udp format. For example, collector@127.0.0.1:5160;transport=udp.

Settings:

<domain name="policy">
<section name="endpoint">
...
<!--
Valid values for Voice Quality (VQ) report publish setting (vq_report_publish):
0--VQ report is not published
1--VQ report is published to the collector at the end of the call-- 
  see the vq_report_collector setting information below
-->
<setting name="vq_report_publish" value="0"/>
<!--
Valid values for Voice Quality (VQ) report collector setting (vq_report_collector):
NULL or Empty--The VQ report is published to the proxy described in the
Connectivity section
FQDN or IP address along with port and transport--
 collector@SipServer.genesyslab.com:5060;transport=udp
-->
<setting name="vq_report_collector" value="collector@SipServer.genesyslab.com:5060;transport=udp"/>
</section>

Endpoint:

The vq_report_publish and vq_report_collector settings can be read from the Endpoint Policy by using the following methods:

GetEndpointPolicy(EndpointPolicyQuery.VqReportCollector);
GetEndpointPolicy(EndpointPolicyQuery.VqReportPublish);

Working with Codec Priorities

Codecs are listed by name in the codecs domain of the configuration file. The enabled section explicitly defines the enabled codecs and their priorities. If the enabled section is not included or if the audio setting has empty values, then the priority is defined as the same order as listed in the codec setting. To disable a codec, remove the codec name from the list.

Important

Media connections between two SIP endpoints (or between SIP Endpoint and Media Server/Gateway) can be successful only when they have a codec that they both support. In tightly controlled environments, it may be possible to limit the number of enabled codecs to only one preferred codec, but it is highly recommended to include the fallback codec(s) like universally supported PCMU and PCMA to the end of the priority list.

The audio setting can include a list of comma-separated audio codec names in the order of their priorities.


Codec parameters can be specified inline as a comma-separated list enclosed in parenthesis after an equal (=) sign.

<setting name="audio" value="g729=(fmtp='annexb=no'),opus=(pt=125),pcmu,pcma"/>


The setting payload_type is an integer. It should be specified for codecs with dynamic payload type, such as iSAC, iLBC, OPUS, and the valid values are between 96 and 127.

The setting fmtp is a string with valid values as defined by RFC3555 for codec g729 and RFC7587 for codec Opus'.

Important

While any parameter defined by RFC7587 can be specified as the value of the fmtp for the Opus codec, SIP Endpoint SDK considers only the maxaveragebitrate parameter, which specifies the maximum average bit rate of a session in bits per second (bit/s). Bit rate values can be from 6000 to 64000 (any value outside the range will be adjusted to the minimum/maximum allowed). Recommended bit-rate values:

  • from 6000 to 8000 to save bandwidth, with voice quality comparable to G.729 codec.
  • 10000 for narrow-band quality (comparable to G.711).
  • 20000 for wide-band quality.

Example

<domain name="codecs">
   <section name="enabled">
      <setting name="audio" value="pcmu,pcma,opus,g722,g729,iSAC/16000,iSAC/32000,iLBC"/>
  </section>
  <section name="PCMU/8000"/>
  <section name="PCMA/8000"/>
  <section name="G722/16000"/>
  <section name="iLBC/8000">
    <setting name="payload_type" value="102"/>
  </section>
  <section name="iSAC/32000">
    <setting name="payload_type" value="104"/>
  </section>
  <section name="iSAC/16000">
    <setting name="payload_type" value="103"/>
  </section>
  <section name="g729/8000">
    <setting name="fmtp" value="annexb=yes"/>
  </section>
  <section name="opus/48000/2">
    <setting name="payload_type" value="120"/>
    <setting name="fmtp" value="maxaveragebitrate=20000"/>
  </section>
  </section>
</domain>


Audio Device Settings

You can configure a headset and other audio input devices using the following parameters in the configuration file:

  • use_headset
  • headset_name
  • audio_in_device
  • audio_out_device


If the SDK cannot access any of the configured audio devices, the SDK uses the default system audio devices.

If use_headset is set to 1, the SDK searches for a headset with configured headset_name. If the SDK does not find the headset, it uses the system defaults.

If use_headset is set to 0, the SDK skips the search for a headset and searches for audio in and audio out devices with names configured by audio_in_device and audio_out_device. If the devices are not found, the SDK uses the system defaults.

The procedure for audio device selection is applied on startup and every time any changes are made to device presence (such as when a new device is plugged in or an existing device is removed)

Important
A device is selected when the device name matches or partially-matches the regular expression configured in the parameters.

Auto-answer and Call Rejection

The auto-answer and call rejection features depend on the use_headset, auto_answer, and reject_session_when_headset_na configuration settings and whether or not audio devices are available. The following tables describe auto-answer and call rejection behavior:


Auto-answer and Call Rejection when use_headset=1

Headset is Available

The SDK considers a headset available if the headset is found by name.

Outgoing calls can be initiated.

auto_answer=1 Incoming calls are answered automatically:
  • As audio if auto_accept_video=0
  • As audio with video if the call has video and auto_accept_video=1
auto_answer=0 Incoming calls are answered manually and the user explicitly selects whether or not video streams should be accepted (using the has_video parameter supplied in the gs_session_info argument)
Headset is Not Available

The SDK decides that no headset is available if a headset was not found by name.

An audio device is still assigned, if possible (that is, if any supported devices are present in the system), using the first available audio input and output devices or the system defaults.

No auto-answer is possible in this sub-case, so the auto_answer setting is not used reject_session_when_headset_na=1
  • Incoming calls are automatically rejected
  • Outgoing calls are blocked
reject_session_when_headset_na=0
  • Incoming calls can be answered manually—it is assumed that the agent will plug the headset in (or use an available non-headset device, if applicable) before answering the call
  • Outgoing calls can be initiated—it is the agent's responsibility to ensure that the appropriate audio devices are available before the call is answered by the remote side

Auto-answer and Call Rejection when use_headset=0

Audio devices are configured using the audio_in_device and audio_out_device settings. If no valid audio_in_device and audio_out_device devices are found, the SDK selects the system defaults. Outgoing calls can be initiated.

Both microphone and speaker are available auto_answer=1 Incoming calls are answered automatically:
  • As audio if auto_accept_video=0
  • As audio with video if the call has video and auto_accept_video=1
auto_answer=0 Incoming calls are answered manually and the user explicitly selects whether or not video streams should be accepted (using the has_video parameter supplied in the gs_session_info argument)
Either microphone or speaker is not available
  • Incoming calls can be answered manually—it is assumed that the agent will plug in the headset (or use an available non-headset device, if applicable) before answering the call
  • Outgoing calls can be initiated—it is the agent's responsibility to ensure that the appropriate audio devices are available before the call is answered by the remote side
No auto-answer is possible in this sub-case, so the auto_answer setting is not used Auto-rejection is not applicable, so the reject_session_when_headset_na setting is not used

Feedback

Comment on this article:

blog comments powered by Disqus
This page was last modified on August 30, 2018, at 23:28.