Here are a few clarifications in regards to QoS and DSCP for the WebRTC softphone. Any posted DSCP values on the Resource Center represent the default values and you can look at your configuration to see which values are being used. It is important to note that DSCP is set by each endpoint, we would prefer that the values are the same in both directions (sent DSCP values and received DSCP values) however there are places to configure these values to ensure that is the case. It is possible to tag in both directions, sending out of the WebRTC phone and receiving from the media service or Edge, as long as the setup is used that supports tagging.
This is an important principle with WebRTC because, as asked, most standard browser lacks the ability to set DSCP values themselves. This is a security precaution in the browser to not allow it to manipulate the raw network traffic. This ability to tag packets is allowed in packaged browser applications such as the Genesys Cloud Desktop Application; hence why only the Desktop application is able to "set" DSCP values for sent traffic.
In Genesys Cloud you have the ability to choose the DSCP values that are used in some places. In both places you have the ability to set the "Signaling" or "Network" value and the "Media" or "RTP" value independently. This means that the control and signaling protocol, which is HTTPS over TCP with TLS for WebRTC, is tagged differently than the media, which is SRTP over UDP. By default media is always "46" (EF) and signaling is "24" (CS3)
Phone Base Settings / Phone SettingsWithin the Phone Base Settings configuration or the Phone settings itself you have the ability to set the DSCP values that the WebRTC phone will send when using the GC Desktop Application.
https://help.mypurecloud.com/articles/configure-the-genesys-cloud-webrtc-phone/WebRTC Phone Trunk SettingsWithin the Phone Trunk Settings configuration you have the ability to set the DSCP values that the remote media service will send back to the WebRTC phone.
https://help.mypurecloud.com/articles/webrtc-phone-trunk-settings/Media Relay / TURNI have observed that when a TURN server is injected into the WebRTC media path that the source DSCP values are not used and rather they are defaulted to the values chosen by the TURN server (possibly "20" CS1). You currently do not have the ability to control this.
Due to the nature of WebRTC signaling relaying on the public Internet, (as well as the Media in BYOC Cloud) there is also concerns about whether DSCP values get manipulated when passed between providers. Internet service is expected to ignore DSCP tags because the Internet is a shared network and there is no way to prioritize traffic in a public forum. However, ignoring tags and manipulating tags are two different things. Upon researching the subject, tag manipulation was done in the past by some service providers, but I cannot find any hard rule about if providers still do it today. There are still benefits to using QoS and DSCP with public network connections and it can still be used as a means of prioritizing inbound and outbound traffic within an enterprise network and leading up to the public demarcation.
------------------------------
Phil Whitener
Genesys - Employees
------------------------------
Original Message:
Sent: 11-18-2020 08:58
From: Shalom Benzaquen
Subject: QoS tagging when using WebRTC through Desktop application
I would like to add to the question, in addition to using the PureDesktop Thick Client that uses connects to a WebRTC phone, the same question of QOS regarding using a Chrome Browser connected to a WebRTC phone.. Thanks
------------------------------
Shalom Benzaquen
Anthem, Inc.
Original Message:
Sent: 11-17-2020 23:56
From: Jeff Hoogkamer
Subject: QoS tagging when using WebRTC through Desktop application
Hi All,
Just looking for a confirmation of my assumptions about QoS tagging when using WebRTC phones for voice traffic through the Desktop application.
From the Prioritizing voice and video traffic help article (extract below), I can see that Voice traffic is assigned a DSCP value of 46 (EF) and WebRTC traffic is assigned 18 (AF21).
Traffic description | Protocol | DSCP value |
---|
Voice traffic | RTP | 46 (EF) |
Video traffic | RTP | 34 (AF41) |
Signalling traffic | SIP | 24 (CS3) |
WebRTC traffic | RTP | 18 (AF21) |
Can I confirm my assumptions:
a) when using WebRTC phones, all traffic is tagged as 18 (AF21) (including the voice traffic over RTP), and
b) only traffic coming from physical phones or Softphones is tagged as "Voice" with 46 (EF)?
Or is my assumption faulty and that only the setup/creation of the WebRTC phone tagged as 18 (AF21), and the actual Voice component is still tagged as 46 (EF)?
Thanks!
#SIP/VolP
#SystemAdministration
#Telephony