[asterisk-bugs] [JIRA] (ASTERISK-29064) Audio Delay on WebRTC Call

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29064?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=251916#comment-251916 ] 

Asterisk Team commented on ASTERISK-29064:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.

> Audio Delay on WebRTC Call
> --------------------------
>
>                 Key: ASTERISK-29064
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29064
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip, Resources/res_rtp_asterisk
>    Affects Versions: 16.9.0
>         Environment: OS: CentOS Linux release 7.7.1908 (Core)
> CPU Cores: 2
> RAM: 4GB
> Client WebRTC Librery: sip.js 0.15.0
> Chrome version:  85.0.4183.83
> Asterisk Behind NAT: yes
> Client behind NAT: yes
>            Reporter: Oliver Rafael Peña
>              Labels: webrtc
>
> Greetings,
> We've deployed asterisk + webrtc in a production enviroment, and since day one we've been having a problem with calls having audio delay  on both ways, this happens on ~10% of the calls, users are more often to identify the problem on outbound calls. The delay goes around 1-10 seconds, it vary. We understand that WebRTC comes with its requirements, and after a few months of research and testing we've done the following trying to solve the problem:
> - Even though we didn't find any fragmentation of DTLS packets on the handshake we decided to activate the pjsip endpoints option "dtls_auto_generate_cert" to discard any problems there.
> - Also we proceed to remove all unnecessary ICE candidates out of the client's end to ease ICE negotiation, along with reducing the ICE timeout to 500ms on the client
> - Also we remove the stunaddr option out of rtp.conf since we know the public IP, therefore removing the stun request out of the process.
> And after all that we still have the issue, hopefully we can clear that out on this thread.
> Based on our most recent testing, we noticed that asterisk is taking a considerable amount of time to initiate the DTLS handshake, roughly 2 seconds after the ICE bindings are done, in the capture file attached you can see that.
> Here are the config files:
> pjsip.endpoint.conf
> {panel}
> [base_webrtc](!)
> type=endpoint
> auth=main_auth
> transport=0.0.0.0-wss
> allow=ulaw,alaw,gsm,g726,g722
> context=from-internal
> dtmf_mode=rfc4733
> mwi_subscribe_replaces_unsolicited=yes
> aggregate_mwi=yes
> use_avpf=yes
> rtcp_mux=yes
> ice_support=yes
> media_use_received_transport=no
> trust_id_inbound=yes
> direct_media=no
> timers=no
> media_encryption_optimistic=yes
> send_pai=yes
> rtp_symmetric=yes
> rewrite_contact=yes
> force_rport=yes
> language=en
> media_encryption=dtls
> dtls_verify=fingerprint
> dtls_auto_generate_cert=yes
> dtls_setup=actpass
> dtls_rekey=0
> webrtc=yes
> rtp_keepalive=2
> [1922285](base_webrtc)
> aors=1922285
> callerid=CallerID <1922285>
> mailboxes=1922285 at device
> {panel}
> rtp.conf
> {panel}
> [general]
> rtpstart=50000
> rtpend=55000
> rtpchecksums=yes
> strictrtp=no
> icesupport=yes
> [ice_host_candidates]
> x.x.x.x => y.y.y.y,include_local_address ; x.x.x.x is the internal IP, y.y.y.y is the external IP
> {panel}
> http.conf
> {panel}
> [general]
> enabled=yes
> enablestatic=no
> bindaddr=::
> bindport=8088
> prefix=
> sessionlimit=1000
> session_inactivity=30000
> session_keep_alive=15000
> tlsenable=yes
> tlsbindaddr=[::]:8089
> tlscertfile=/etc/asterisk/keys/cert.crt
> tlsprivatekey=/etc/asterisk/keys/keyl.key
> {panel}



--
This message was sent by Atlassian JIRA
(v6.2#6252)