rsmp Section
- allow-anonymous-user
- allow-ipv6
- codecs
- domain-whitelist
- enable-https
- enable-transcoding
- http-port
- http-trace
- https-cert
- https-cert-key
- https-trusted-ca
- reporting-service-type
- rtp-address
- rtp-trace-level
- sip-added-codecs
- sip-address
- sip-disallowed-codecs
- sip-no-avpf
- sip-no-rtcpfb
- sip-port
- sip-preferred-ipversion
- sip-proxy
- sip-register
- sip-rtp-max-port
- sip-rtp-min-port
- sip-srtp-mode
- sip-tls-cert
- sip-tls-cert-key
- sip-tls-port
- sip-tls-trusted-ca
- stun-server
- turn-passwd
- turn-relay-type
- turn-server
- turn-user
- web-added-codecs
- web-disallowed-codecs
- web-dtls-certificate
- web-dtls-cipherlist
- web-dtls-keypassword
- web-dtls-privatekey
- web-enable-dtls
- web-ice-addresses
- web-media-bundle
- web-nack-enabled
- web-pli-always
- web-pli-mintime
- web-rtcp-mux
- web-rtp-max-port
- web-rtp-min-port
allow-anonymous-user
Default Value: true
Valid Values:
Changes Take Effect: At start or restart
Set this to true (default) to enable anonymous users to sign-in to the WebRTC Gateway. If set to false then only registered users for SIP Server can sign-in.
allow-ipv6
Default Value: false
Valid Values:
Changes Take Effect: At start or restart
Controls whether IPv6 is allowed in the WebRTC Gateway.
codecs
Default Value: (pcmu,pcma,opus,g729,telephone-event=126,vp8=100,h264=(pt=108,fmtp="[profile-level-id=42000B;packetization-mode=1]"))
Valid Values:
Changes Take Effect: At start or restart
Codecs that are not listed here will not be used in an offer or answer. The codec's clock rate (in Hz) can also be specified with the name following a '/'. The codecs currently supported are: pcmu (G.711 mu-Law), pcma (G.711 A-Law), g722, g723 (G.723.1), g729 (G.729/a/b), iLBC, iSAC/16000, iSAC/32000, vp8, h264, telephone-event and opus (non-transcoding case). A default payload type number can be specified using the format name=<pt>, or name=(pt=<pt>). The latter format needs to be used if an fmtp is to be specified, which will be specified as fmtp=<fmtp>. A comma is used as a separator between the different values. All or part of the fmtp value can be enclosed within square brackets, where those brackets will be removed when used in an offer, and in the case of an answer, the brackets and the content will be replaced by the fmtp value from the remote offer.
domain-whitelist
Default Value:
Valid Values:
Changes Take Effect: At start or restart
A list of comma separated domain values that are used to match the domain in the Origin header of HTTP requests. If there is no match, a "403 Forbidden" error will be returned, although an empty (default) whitelist will disallow this checking altogether. Each domain entry may have wildcard character '*' to specify arbitrary scheme, port or sub-domains. Here is a sample whitelist: "https://my.foo.com:8081, http://*.foo2.com, *://*.sub.foo3.com:*, *foo4.com". A '*' at start would match HTTP or HTTPS. If it is immediately followed by a domain name or if a '*' comes after "://" before the domain name, then any sub-domain with the specified name will match; otherwise, domain names will have to exactly match. Also, ":*" at the end would match any port. If no port specified, however, then the default HTTP port 80 is assumed.
enable-https
Default Value: false
Valid Values:
Changes Take Effect: At start or restart
Enables HTTPS
enable-transcoding
Default Value: false
Valid Values:
Changes Take Effect: At start or restart
Transcoding of audio and/or video between the SIP and Web sides is enabled this value is set to true. Otherwise, transcoding will be disabled. When enabled, transcoding will be activated for a media type, only when there is no common codec negotiated between the sides, or when a codec sent by one side is not supported by the other side.
http-port
Default Value: 8086
Valid Values:
Changes Take Effect: At start or restart
HTTP or HTTPS port
http-trace
Default Value: false
Valid Values:
Changes Take Effect: At start or restart
Traces HTTP requests and responses
https-cert
Default Value:
Valid Values:
Changes Take Effect: At start or restart
For Windows, the thumbprint obtained from the user certificate generated for the host. For Linux, the fullpath of the host certificate file (.pem).
https-cert-key
Default Value:
Valid Values:
Changes Take Effect: At start or restart
Applicable for Linux only. The fullpath of the host private key file (.pem).
https-trusted-ca
Default Value:
Valid Values:
Changes Take Effect: At start or restart
Applicable for Linux only. The fullpath of the Certificate Authority file (.pem).
reporting-service-type
Default Value: WebRTC
Valid Values:
Changes Take Effect: At start or restart
SIP calls are reported out of the box when SIP Server and ICON are configured. When this parameter is set, the service_type key-value pair is sent to SIP Server and then reported to ICON. This allows the reports for the WebRTC service to be filtered based on the service type specified here. To disable the sending of a service type set this parameter value to "none".
rtp-address
Default Value:
Valid Values:
Changes Take Effect: At start or restart
Allows for configuration of a specific IP address for SDP c= line. If not set, the stack will attempt to detect the IP address automatically. This is useful for AWS instances or multi-homed hosts. For example, in an AWS instance you can set this to the elastic-IP. This setting applies to the SIP side only.
rtp-trace-level
Default Value: 1
Valid Values:
Changes Take Effect: At start or restart
The RTP trace level controls how many packets are printed into the log.
sip-added-codecs
Default Value: (vp8,h264)
Valid Values:
Changes Take Effect: At start or restart
When transcoding is enabled, codecs from this list will be appended to the codec list for offers to a SIP endpoint, after removing any codecs that are already in the original offer. If not specified here, the pt and the fmtp values will be used from the list specified in the codecs option. Note that at least one video codec should be specified, and this codec should most likely be supported by the SIP side. Otherwise, the call may fail even if transcoding is supported. For example, if the Web side offers only VP8, and the SIP side only supports H.264, sip-added-codecs will need to contain h264. If a common audio codec is disallowed on one side, then it should be added to the other side for similar reasons. For video upgrade case on the SIP side, with REFER for example, it is good to have VP8 too.
sip-address
Default Value:
Valid Values:
Changes Take Effect: At start or restart
Allows for configuration of a specific IP address for SIP Via or Contact. If not set, the stack will attempt to detect the IP address automatically. This is useful for AWS instances or multi-homed hosts. For example, in an AWS instance you can set this to the elastic-IP.
sip-disallowed-codecs
Default Value:
Valid Values:
Changes Take Effect: At start or restart
Disallowed codecs for the SIP side. An offer or answer to the SIP side may not use any of these codecs.
sip-no-avpf
Default Value: true
Valid Values:
Changes Take Effect: At start or restart
Set this to true in order not to negotiate AVPF in SDP on the SIP side (RFC4585). This is necessary to work with SIP endpoints that do not support AVPF. Note that regardless of the value of this option, if sip-no-rtcpfb = false, RTCP feedback messages will be forwarded to the SIP side. These settings are useful for a Chrome-to-Chrome call.
sip-no-rtcpfb
Default Value: false
Valid Values:
Changes Take Effect: At start or restart
If set to false, RTCP feedback messages sent by a WebRTC client in accordance with RFC4585 will be forwarded to the corresponding SIP endpoint in a call. A true value will disable this. Note that even though endpoints should ignore RTCP packets of unknown types, some may have issues with this.
sip-port
Default Value: 5066
Valid Values:
Changes Take Effect: At start or restart
SIP Port
sip-preferred-ipversion
Default Value: ipv4
Valid Values:
Changes Take Effect: At start or restart
Preferred IP version to be used for SIP.
sip-proxy
Default Value: 127.0.0.1
Valid Values:
Changes Take Effect: At start or restart
The SIP Proxy and Registrar to be used by the WebRTC Gateway. In all scenarios a Genesys SIP Server is specified as the proxy and registrar.
sip-register
Default Value:
Valid Values:
Changes Take Effect: At start or restart
The list of DNs configured in SIP Server for registration.
sip-rtp-max-port
Default Value: 9999
Valid Values:
Changes Take Effect: At start or restart
UDP port range for SIP-side RTP connection.
sip-rtp-min-port
Default Value: 9000
Valid Values:
Changes Take Effect: At start or restart
UDP port range for SIP-side RTP connection.
sip-srtp-mode
Default Value: none
Valid Values:
Changes Take Effect: At start or restart
SRTP mode that is to be used in SDP negotiation on the SIP side.
sip-tls-cert
Default Value:
Valid Values:
Changes Take Effect: At start or restart
For Windows, the thumbprint obtained from the user certificate generated for the host. For Linux, the fullpath of the host certificate file (.pem)
sip-tls-cert-key
Default Value:
Valid Values:
Changes Take Effect: At start or restart
Applicable for Linux only. The fullpath of the host private key file (.pem).
sip-tls-port
Default Value: 0
Valid Values:
Changes Take Effect: At start or restart
SIP TLS Port. To disable TLS transport for SIP traffic altogether, set to 0.
sip-tls-trusted-ca
Default Value:
Valid Values:
Changes Take Effect: At start or restart
Applicable for Linux only. The fullpath of the Certificate Authority file (.pem).
stun-server
Default Value:
Valid Values:
Changes Take Effect: At start or restart
Optional STUN server specification (port may be omitted, if default STUN port 3478 is used). Only local addresses are gathered when STUN or TURN is not configured.
turn-passwd
Default Value:
Valid Values:
Changes Take Effect: At start or restart
The TURN password to use for the allocation.
turn-relay-type
Default Value: 0
Valid Values:
Changes Take Effect: At start or restart
The type of relay to use. TCP(1) and UDP(0) are supported; TLS is not supported. The default is UDP.
turn-server
Default Value:
Valid Values:
Changes Take Effect: At start or restart
Optional TURN server specification (port may be omitted, if default TURN port 3478 is used). Only local addresses are gathered when STUN or TURN is not configured.
turn-user
Default Value:
Valid Values:
Changes Take Effect: At start or restart
The TURN username to use for the allocation.
web-added-codecs
Default Value: (pcmu,vp8)
Valid Values:
Changes Take Effect: At start or restart
When transcoding is enabled, codecs from this list will be appended to the codec list for offers to a WebRTC endpoint, after removing any codecs that are already in the original offer. The other comments for sip-added-codecs are applicable here as well.
web-disallowed-codecs
Default Value:
Valid Values:
Changes Take Effect: At start or restart
Disallowed codecs for the WebRTC side. An offer or answer to the Web side may not use any of these codecs.
web-dtls-certificate
Default Value: ../config/x509_certificate.pem
Valid Values:
Changes Take Effect: At start or restart
Path of the X.509 certificate file to be used with Web-side DTLS. This file can also contain the private key for the certificate, in which case web-dtls-privatekey does not need to be set. The certificate file is mandatory for DTLS to work. The default certificate already contains the private key.
web-dtls-cipherlist
Default Value:
Valid Values:
Changes Take Effect: At start or restart
A list of cipher strings to be used with DTLS on the Web side. For information on the format, see http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS. The default cipher string should work well.
web-dtls-keypassword
Default Value:
Valid Values:
Changes Take Effect: At start or restart
The password for the private key specified using web-dtls-privatekey, if used.
web-dtls-privatekey
Default Value:
Valid Values:
Changes Take Effect: At start or restart
Path of the private key file for the certificate specified in web-dtls-certificate. This parameter is not necessary if the certificate file also contains the private key.
web-enable-dtls
Default Value: true
Valid Values:
Changes Take Effect: At start or restart
When this is set to true, DTLS-SRTP (RFC 5763) will be enabled on the Web side. When enabled, it will be signalled in an SDP offer sent by the gateway using the fingerprint attributes, though there will also be crypto attributes in SDP for SDES-SRTP (RFC 4568) support. When an offer or answer comes in with only crypto attributes, then SDES-SRTP will still be supported. When this is set to false, only SDES-SRTP will be supported.
web-ice-addresses
Default Value:
Valid Values:
Changes Take Effect: At start or restart
Allows for configuration of a local IP address' list to be used with ICE on the Web/ROAP side. Comma is the delimiter, and each IP address could be IPv4 or IPv6 (no need for square brackets). These addresses are used by ICE to gather host candidates.
web-media-bundle
Default Value: true
Valid Values:
Changes Take Effect: At start or restart
Set this to true to enable media bundling on the ROAP side (see http://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-03). When enabled, it will be signalled in an SDP offer sent by the gateway, and it will be accepted from an inbound SDP offer. If both sides agree, then the same media port will be used for both audio and video. Set this to false if media bundling is not to be used.
web-nack-enabled
Default Value: true
Valid Values:
Changes Take Effect: At start or restart
Set this to true (default) to enable RTCP NACK (transport layer) feedback messages as per RFC4585. Set this to false to disable this feature. The minimum time between two NACK messages is currently restricted to one second.
web-pli-always
Default Value: true
Valid Values:
Changes Take Effect: At start or restart
If this parameter is set to true and web-pli-mintime is nonzero, RTCP PLI feedback messages (RFC4585) will be sent on a Web-side video leg at every web-pli-mintime interval, regardless of transcoding or packet losses.
web-pli-mintime
Default Value: 1000
Valid Values: The parameter must be an integer.
Changes Take Effect: At start or restart
The minimum time period, in milliseconds, between two RTCP PLI feedback messages (RFC4585) that can be sent on a Web-side video leg. If this value is 0, PLI transmission is disabled. The actual time between two PLI messages depends on various things: if web-pli-always is true, one message will be sent every web-pli-mintime milliseconds. Otherwise, if transcoding is on, a message will be sent when the number of lost packets during web-pli-mintime exceed a specific threshold.
web-rtcp-mux
Default Value: true
Valid Values:
Changes Take Effect: At start or restart
Set this to true to enable rtcp-mux on the ROAP side, as per RFC 5761. When enabled, it will be signalled in an SDP offer sent by the gateway, and it will be accepted from an inbound SDP offer. If both sides agree, then the same port will be used for both RTP and RTCP. Set this to false if rtcp-mux is not to be used. Note: If web-rtcp-mux is false, then web-media-bundle cannot be true, as it would not make sense.
web-rtp-max-port
Default Value: 36999
Valid Values:
Changes Take Effect: At start or restart
Maximum UDP port value for ICE (ROAP-side RTP connection). If not specified or zero, then ICE agent is free to select ports by itself (ports in the recommended range of 36000 through 36999 are opened in both Genesys and Amazon cloud firewalls).
web-rtp-min-port
Default Value: 36000
Valid Values:
Changes Take Effect: At start or restart
Minimum UDP port value for ICE (ROAP-side RTP connection). If not specified or zero, then ICE agent is free to select ports by itself (ports in the recommended range of 36000 through 36999 are opened in both Genesys and Amazon cloud firewalls).