[asterisk-bugs] [JIRA] (ASTERISK-29064) Audio Delay on WebRTC Call
Oliver Rafael Peña (JIRA)
noreply at issues.asterisk.org
Wed Sep 2 18:40:43 CDT 2020
[ https://issues.asterisk.org/jira/browse/ASTERISK-29064?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Oliver Rafael Peña updated ASTERISK-29064:
------------------------------------------
Description:
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}
was:
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}
> 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
> Attachments: Audio Delay Call - udp capture.pcapng, Audio Delay SIP Messages.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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