[asterisk-bugs] [JIRA] (ASTERISK-27826) No sound in some calls (webrtc)
Mikhail Ivanov (JIRA)
noreply at issues.asterisk.org
Thu Apr 26 13:11:51 CDT 2018
[ https://issues.asterisk.org/jira/browse/ASTERISK-27826?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=243190#comment-243190 ]
Mikhail Ivanov commented on ASTERISK-27826:
-------------------------------------------
yes
alaw only
asterisk*CLI> pjsip show endpoint 5191
Endpoint: <Endpoint/CID.....................................> <State.....> <Channels.>
I/OAuth: <AuthId/UserName...........................................................>
Aor: <Aor............................................> <MaxContact>
Contact: <Aor/ContactUri..........................> <Hash....> <Status> <RTT(ms)..>
Transport: <TransportId........> <Type> <cos> <tos> <BindAddress..................>
Identify: <Identify/Endpoint.........................................................>
Match: <criteria.........................>
Channel: <ChannelId......................................> <State.....> <Time.....>
Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......>
==========================================================================================
Endpoint: 5191/5191 Not in use 0 of inf
OutAuth: 5191/5191
InAuth: 5191/5191
Aor: 5191 1
Contact: 5191/sip:qv1njbdk at 127.0.0.1:57521;transpor e27cb64e55 Unknown nan
ParameterName : ParameterValue
====================================================================
100rel : yes
accountcode :
acl :
aggregate_mwi : true
allow : (alaw)
allow_overlap : true
allow_subscribe : true
allow_transfer : true
aors : 5191
asymmetric_rtp_codec : false
auth : 5191
bind_rtp_to_media_address : false
bundle : true
call_group :
callerid : "М. Иванов" <5191>
callerid_privacy : allowed_not_screened
callerid_tag :
connected_line_method : invite
contact_acl :
context : default
cos_audio : 0
cos_video : 0
device_state_busy_at : 0
direct_media : false
direct_media_glare_mitigation : none
direct_media_method : invite
disable_direct_media_on_nat : false
dtls_auto_generate_cert : No
dtls_ca_file :
dtls_ca_path :
dtls_cert_file : /etc/asterisk/keys/asterisk.pem
dtls_cipher :
dtls_fingerprint : SHA-256
dtls_private_key : /etc/asterisk/keys/asterisk.key
dtls_rekey : 0
dtls_setup : actpass
dtls_verify : Yes
dtmf_mode : rfc4733
fax_detect : false
fax_detect_timeout : 0
force_avp : false
force_rport : true
from_domain :
from_user :
g726_non_standard : false
ice_support : true
identify_by : username,ip
inband_progress : false
incoming_mwi_mailbox :
language :
mailboxes :
max_audio_streams : 1
max_video_streams : 1
media_address :
media_encryption : dtls
media_encryption_optimistic : false
media_use_received_transport : true
message_context :
moh_passthrough : false
moh_suggest : default
mwi_from_user :
mwi_subscribe_replaces_unsolicited : false
named_call_group :
named_pickup_group :
notify_early_inuse_ringing : false
one_touch_recording : false
outbound_auth : 5191
outbound_proxy :
pickup_group :
preferred_codec_only : false
record_off_feature : automixmon
record_on_feature : automixmon
refer_blind_progress : true
rewrite_contact : false
rpid_immediate : false
rtcp_mux : true
rtp_engine : asterisk
rtp_ipv6 : false
rtp_keepalive : 0
rtp_symmetric : true
rtp_timeout : 0
rtp_timeout_hold : 0
sdp_owner : -
sdp_session : Asterisk
send_diversion : true
send_pai : false
send_rpid : false
set_var :
srtp_tag_32 : false
sub_min_expiry : 0
subscribe_context :
t38_udptl : false
t38_udptl_ec : none
t38_udptl_ipv6 : false
t38_udptl_maxdatagram : 0
t38_udptl_nat : false
timers : yes
timers_min_se : 90
timers_sess_expires : 1800
tone_zone :
tos_audio : 0
tos_video : 0
transport :
trust_id_inbound : false
trust_id_outbound : false
use_avpf : true
use_ptime : false
user_eq_phone : false
voicemail_extension :
webrtc : yes
asterisk*CLI>
SIP and RTP I'll try to capture tomorrow....
> No sound in some calls (webrtc)
> -------------------------------
>
> Key: ASTERISK-27826
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-27826
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Resources/res_rtp_asterisk
> Affects Versions: 15.3.0
> Reporter: Mikhail Ivanov
> Assignee: Mikhail Ivanov
> Labels: fax, pjsip, webrtc
>
> I have a problem with incoming (may be with outgoing too, not sure) calls to WebRTC clients (based on jssip.net library)
> Sometimes (2-5% of all incoming calls) I have no sound (on both sides) on incoming calls.
> RTP is going fine in both sides (local network)
> If I turn on mixMonitor on Asterisk, I can see only noise in call (looks like a problem with srtp keys, but not sure)
> https://www.dropbox.com/s/41nmwqhg0chcwl7/cf626000ac4601445d6cee3cd909188d.mp3?dl=1
> Asterisk 15.3.0, JsSIP 3.2.8, tested in Chrome, Chromium and Firefox
> If I turn off rtp encryption
> webrtc = no
> rtcp_mux = yes
> use_avpf = yes
> ice_support = yes
> media_encryption = no
> and
> --disable-webrtc-encryption in crome
> everything is fine, yes, it's workaround but not a solution
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