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

[Benjamin Keith Ford (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-29064?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Benjamin Keith Ford updated ASTERISK-29064:
-------------------------------------------

    Assignee: Oliver Rafael Peña  (was: Unassigned)
      Status: Waiting for Feedback  (was: Triage)

Can you get the same logs but with much higher debug levels (i.e., 9)? Also, can you provide some clarification on the IP addresses so we can follow the conversation between them?

> res_rtp_asterisk: 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
>            Assignee: Oliver Rafael Peña
>              Labels: webrtc
>         Attachments: audio delay call - 15 seconds no audio.pcapng, audio delay call - 15 seconds no audio SIP Messages.txt, Audio Delay Call - udp capture.pcapng, Audio Delay SIP Messages.txt, debug_15seconds_noaudio_call.txt
>
>
> 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. The users perceived around 4-5 seconds of none audio at the beginning of this call.
> 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}



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