WebRtc : no sound

I’m using asterisk 16.2.1 (chan_sip) on Debian 10.10.
I’ve successfully manage to establish a websocket (vujs app with JsSIP library + public signed cert) connexion and a call (MusicOnhold).
I’m observing RTP stream on asterisk and on my client computer (wireshark) but unfortunately no sound on my computer (I should head the moh sound, setup successfully tested with direct SIP softphone)
Client browser connect to asterisk through a corporate routed network (no NAT).
If you have some direction to debug this issue …
Thank you very much



exten => _X.,1,NoOP(from webrtc)
same => n, Ringing()
same => n, Wait(2)
same => n, Answer()
same => n, MusicOnHold()

;;sip show settings
Global Settings:

UDP Bindaddress:
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
RTP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: No
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Path support : No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u2
SDP Session Name: Asterisk PBX 16.2.1~dfsg-1+deb10u2
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: 4294967295
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No

Network QoS Settings:

IP ToS RTP audio: CS0
IP ToS RTP video: CS0
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No

Network Settings:

SIP address remapping: Disabled, no localnet list
Externaddr: (null)
Externrefresh: 10

Global Signalling Settings:

Codecs: (ulaw|alaw|gsm|h263)
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Sub. min duration 60 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Outbound reg. retry 403:No
Notify ringing state: Yes
Include CID: No
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy:
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70

Default Settings:

Allowed transports: UDP,WSS
Outbound transport: UDP
Context: public
Record on feature: automon
Record off feature: automon
Force rport: Auto (No)
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: No
Tone zone:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk
RTCP Multiplexing: No

;; dpkg -l |grep asterisk
ii asterisk 1:16.2.1~dfsg-1+deb10u2 amd64 Open Source Private Branch Exchange (PBX)
ii asterisk-config 1:16.2.1~dfsg-1+deb10u2 all Configuration files for Asterisk
ii asterisk-core-sounds-en 1.6.1-1 all asterisk PBX sound files - US English
ii asterisk-core-sounds-en-gsm 1.6.1-1 all asterisk PBX sound files - en-us/gsm
ii asterisk-modules 1:16.2.1~dfsg-1+deb10u2 amd64 loadable modules for the Asterisk PBX
ii asterisk-moh-opsound-g722 2.03-1 all asterisk extra sound files - English/g722
ii asterisk-moh-opsound-wav 2.03-1 all asterisk extra sound files - English/wav
ii asterisk-opus 13.7+20171009-2 amd64 opus module for Asterisk

;;; asterisk -rx “http show status”
HTTP Server Status:
Server: Asterisk
Server Enabled and Bound to

HTTPS Server Enabled and Bound to

Enabled URI’s:
/httpstatus => Asterisk HTTP General Status
/phoneprov/… => Asterisk HTTP Phone Provisioning Tool
/amanager => HTML Manager Event Interface w/Digest authentication
/arawman => Raw HTTP Manager Event Interface w/Digest authentication
/manager => HTML Manager Event Interface
/rawman => Raw HTTP Manager Event Interface
/static/… => Asterisk HTTP Static Delivery
/amxml => XML Manager Event Interface w/Digest authentication
/mxml => XML Manager Event Interface
/ari/… => Asterisk RESTful API
/ws => Asterisk HTTP WebSocket

Enabled Redirects:

No audio with webrtc is usually always ICE related, focus your efforts there.

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