Asterisk 18 with webrtc one way audio please hepl me

ahmedna · January 3, 2023, 7:53am

i have installed asterisk 18 on ubuntu 20.04 and jssip.js WebRTC client but work only one way audio the rtp packegs form asterisk to jssip client is work but jssip client to asterisk that is sillent side

sip.conf

[general]
bindport=5060
srvlookup=yes
nat=no



localnet=10.12.39.0/24



context=default
realm=xx.xxx.xx 
transport=udp,ws,wss
directmedia=no ;nonat
mohsuggest=default
parkinglot=default
;allowguest=no
;alwaysauthreject=yes
videosupport=no
maxcallbitrate=1024
ignoreregexpire=no
useragent= HRMBPX03

allowsubscribe=yes
notifyhold=yes
notifyringing=yes
callcounter=yes

prematuremedia=no ;
progressinband=never ; yes|no|never

tos_sip=af42
tos_audio=ef
cos_sip=3
cos_audio=5

; rtptimeout=120
; rtpkeepalive=10

jbenable=yes
jbforce=no
jbmaxsize=200
jbresyncthreshold=1000
jbimpl=fixed
jblog=no

tcpenable=no
tlsenable=no
websocket_enabled=yes
encryption=no

disallow=all
allow=gsm
allow=alaw
allow=ulaw
allow=ilbc

rtcachefriends=yes
rtautoclear=no
rtsavesysname=yes
rtupdate=yes








[basic](!)
type=friend
qualify=yes
context=default
subscribecontext=subscriptions
host=dynamic
directmedia=no
nat=yes
dtmfmode=rfc2833
disallow=all
videosupport=yes

[phones](!)
transport=udp
allow=ulaw,alaw,g722,gsm,vp9,vp8,h264


[webrtc](!)
transport=wss
allow=gsm,ulaw,alaw,g722,opus,vp9,vp8,h264
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
rtcp_mux=yes
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt 
dtlssetup=actpass




[444]
type=friend
host=dynamic
context=Hormuud_49
username=444
secret=mysecrate
nat=no
directmedia=no


[666]
type=friend
host=dynamic
context=default
username=666
secret=mysecrate
nat=force_rport,comedia,auto_force_rport,auto_comedia
directmedia=no

[333](basic,webrtc)
username=333
secret=mysecrate

rtp.conf


[general]

;

; RTP start and RTP end configure start and end addresses

;

; Defaults are rtpstart=5000 and rtpend=31000

;

rtpstart=10000

rtpend=20000

; icesupport=true

icesupport=yes```

ahmedna · January 3, 2023, 7:53am

sip debug


<--- SIP read from UDP:10.11.63.77:5068 --->
INVITE sip:2059@10.12.39.223;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.11.63.77:5068;branch=z9hG4bKfx7diivx8v4d8f7uww76ini8t;X-DispMsg=1407
Route: <sip:10.12.39.223:5060;transport=udp;lr>
Call-ID: ptwfiio47dtot7ofju4tf6fvnvdvvj4d@10.18.5.64
From: "611693419"<sip:611693419@10.11.63.77;transport=udp;user=phone>;tag=ivvxtjvv-CC-1004-OFC-71
To: "2059"<sip:2059@10.12.39.223;transport=udp;user=phone>
CSeq: 1 INVITE
P-Access-Network-Info: GEN-ACCESS;"area-number=+2521"
Max-Forwards: 70
Contact: <sip:10.11.63.77:5060>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,UPDATE
P-Asserted-Identity: <tel:611693419>
P-Early-Media: supported
Supported: 100rel,timer,histinfo,precondition
Min-SE: 90
Session-Expires: 1800;refresher=uac
Content-Length: 732
Content-Type: multipart/mixed;boundary=ssboundary

--ssboundary
Content-Length: 512
Content-Type: application/sdp

v=0
o=HuaweiSoftx3000 1130124119 1130124120 IN IP4 10.11.63.77
s=SipCall
c=IN IP4 10.11.5.200
t=0 0
m=audio 4392 RTP/AVP 108 102 8 0 18 116
a=rtpmap:108 AMR/8000
a=fmtp:108 mode-change-neighbor=1;mode-change-period=2
a=rtpmap:102 AMR/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:116 telephone-event/8000
a=ptime:20
a=maxptime:20
a=curr:qos local sendrecv
a=curr:qos remote none
a=des:qos mandatory local sendrecv
a=des:qos optional remote sendrecv
a=3gOoBTC

--ssboundary
Content-Length: 49
Content-Type: application/isup;version=itu-t92+

`
<------------->
--- (18 headers 30 lines) ---
Sending to 10.11.63.77:5068 (no NAT)
Sending to 10.11.63.77:5068 (no NAT)
Using INVITE request as basis request - ptwfiio47dtot7ofju4tf6fvnvdvvj4d@10.18.5.64
No matching peer for '611693419' from '10.11.63.77:5068'
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Got SDP version 1130124120 and unique parts [HuaweiSoftx3000 1130124119 IN IP4 10.11.63.77]
Found RTP audio format 108
Found RTP audio format 102
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 116
Found unknown media description format AMR for ID 108
Found unknown media description format AMR for ID 102
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 116
Capabilities: us - (gsm|alaw|ulaw|ilbc), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
       > 0x7f0ba80639c0 -- Strict RTP learning after remote address set to: 10.11.5.200:4392
Peer audio RTP is at port 10.11.5.200:4392
Looking for 2059 in default (domain 10.12.39.223)
[Jan  3 10:27:49] ERROR[356612]: cdr.c:3357 ast_cdr_getvar: Unable to find CDR for channel SIP/10.11.63.77-0000004b
[Jan  3 10:27:49] ERROR[356612]: cdr.c:3357 ast_cdr_getvar: Unable to find CDR for channel SIP/10.11.63.77-0000004b
[Jan  3 10:27:49] ERROR[356612]: cdr.c:3357 ast_cdr_getvar: Unable to find CDR for channel SIP/10.11.63.77-0000004b
sip_route_dump: route/path hop: <sip:10.11.63.77:5060>

<--- Transmitting (no NAT) to 10.11.63.77:5068 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.11.63.77:5068;branch=z9hG4bKfx7diivx8v4d8f7uww76ini8t;X-DispMsg=1407;received=10.11.63.77
From: "611693419"<sip:611693419@10.11.63.77;transport=udp;user=phone>;tag=ivvxtjvv-CC-1004-OFC-71
To: "2059"<sip:2059@10.12.39.223;transport=udp;user=phone>
Call-ID: ptwfiio47dtot7ofju4tf6fvnvdvvj4d@10.18.5.64
CSeq: 1 INVITE
Server: MyPBXServer9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:2059@10.12.39.223:5060>
Content-Length: 0


<------------>
Audio is at 11304
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 10.11.63.77:5068 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.11.63.77:5068;branch=z9hG4bKfx7diivx8v4d8f7uww76ini8t;X-DispMsg=1407;received=10.11.63.77
From: "611693419"<sip:611693419@10.11.63.77;transport=udp;user=phone>;tag=ivvxtjvv-CC-1004-OFC-71
To: "2059"<sip:2059@10.12.39.223;transport=udp;user=phone>;tag=as056f1cf4
Call-ID: ptwfiio47dtot7ofju4tf6fvnvdvvj4d@10.18.5.64
CSeq: 1 INVITE
Server: MyPBXServer9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:2059@10.12.39.223:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 276

v=0
o=root 1725158103 1725158103 IN IP4 10.12.39.223
s=Asterisk PBX 18.15.0
c=IN IP4 10.12.39.223
t=0 0
m=audio 11304 RTP/AVP 8 0 116
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>

<--- SIP read from UDP:10.11.63.77:5068 --->
ACK sip:2059@10.12.39.223:5060 SIP/2.0
Via: SIP/2.0/UDP 10.11.63.77:5068;branch=z9hG4bKp4t4jpunfjunv8fdnijuovvfo;X-DispMsg=1407
Call-ID: ptwfiio47dtot7ofju4tf6fvnvdvvj4d@10.18.5.64
From: "611693419"<sip:611693419@10.11.63.77;transport=udp;user=phone>;tag=ivvxtjvv-CC-1004-OFC-71
To: "2059"<sip:2059@10.12.39.223;transport=udp;user=phone>;tag=as056f1cf4
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:10.11.63.77:5068 --->
INVITE sip:2059@10.12.39.223:5060 SIP/2.0
Via: SIP/2.0/UDP 10.11.63.77:5068;branch=z9hG4bKvpnxi7d4pfpvo8n4tudidtj6u;X-DispMsg=1407
Call-ID: ptwfiio47dtot7ofju4tf6fvnvdvvj4d@10.18.5.64
From: "611693419"<sip:611693419@10.11.63.77;transport=udp;user=phone>;tag=ivvxtjvv-CC-1004-OFC-71
To: "2059"<sip:2059@10.12.39.223;transport=udp;user=phone>;tag=as056f1cf4
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <sip:10.11.63.77:5060>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE
Supported: timer
Content-Length: 203
Content-Type: application/sdp

v=0
o=HuaweiSoftx3000 1130124119 1130124121 IN IP4 10.11.63.77
s=SipCall
c=IN IP4 10.11.5.200
t=0 0
m=audio 4392 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=ptime:20
<------------->
--- (12 headers 9 lines) ---
Sending to 10.11.63.77:5068 (no NAT)
Comparing SDP version 1130124120 -> 1130124121 and unique parts [HuaweiSoftx3000 1130124119 IN IP4 10.11.63.77] -> [HuaweiSoftx3000 1130124119 IN IP4 10.11.63.77]
Found RTP audio format 8
Found RTP audio format 116
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 116
Capabilities: us - (gsm|alaw|ulaw|ilbc), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
       > 0x7f0ba80639c0 -- Strict RTP learning after remote address set to: 10.11.5.200:4392
Peer audio RTP is at port 10.11.5.200:4392

<--- Transmitting (no NAT) to 10.11.63.77:5068 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.11.63.77:5068;branch=z9hG4bKvpnxi7d4pfpvo8n4tudidtj6u;X-DispMsg=1407;received=10.11.63.77
From: "611693419"<sip:611693419@10.11.63.77;transport=udp;user=phone>;tag=ivvxtjvv-CC-1004-OFC-71
To: "2059"<sip:2059@10.12.39.223;transport=udp;user=phone>;tag=as056f1cf4
Call-ID: ptwfiio47dtot7ofju4tf6fvnvdvvj4d@10.18.5.64
CSeq: 2 INVITE
Server: MyPBXServer9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:2059@10.12.39.223:5060>
Content-Length: 0


<------------>
Audio is at 11304
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 10.11.63.77:5068 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.11.63.77:5068;branch=z9hG4bKvpnxi7d4pfpvo8n4tudidtj6u;X-DispMsg=1407;received=10.11.63.77
From: "611693419"<sip:611693419@10.11.63.77;transport=udp;user=phone>;tag=ivvxtjvv-CC-1004-OFC-71
To: "2059"<sip:2059@10.12.39.223;transport=udp;user=phone>;tag=as056f1cf4
Call-ID: ptwfiio47dtot7ofju4tf6fvnvdvvj4d@10.18.5.64
CSeq: 2 INVITE
Server: MyPBXServer9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:2059@10.12.39.223:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 252

v=0
o=root 1725158103 1725158104 IN IP4 10.12.39.223
s=Asterisk PBX 18.15.0
c=IN IP4 10.12.39.223
t=0 0
m=audio 11304 RTP/AVP 8 116
a=rtpmap:8 PCMA/8000
a=rtpmap:116 telephone-event/8000
a=fmtp:116 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>

<--- SIP read from UDP:10.11.63.77:5068 --->
ACK sip:2059@10.12.39.223:5060 SIP/2.0
Via: SIP/2.0/UDP 10.11.63.77:5068;branch=z9hG4bK8tj8v87ij66podn7x6f68ioi7;X-DispMsg=1407
Call-ID: ptwfiio47dtot7ofju4tf6fvnvdvvj4d@10.18.5.64
From: "611693419"<sip:611693419@10.11.63.77;transport=udp;user=phone>;tag=ivvxtjvv-CC-1004-OFC-71
To: "2059"<sip:2059@10.12.39.223;transport=udp;user=phone>;tag=as056f1cf4
CSeq: 2 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
       > 0x7f0ba80639c0 -- Strict RTP switching to RTP target address 10.11.5.200:4392 as source
 /home/callCenter/full/callcenter_v1.py: goToQueue
    -- AGI Script Executing Application: (MixMonitor) Options: (/home/callrecords/49/1672730869.126.wav,b)
  == Begin MixMonitor Recording SIP/10.11.63.77-0000004b
    -- AGI Script Executing Application: (Queue) Options: (49_ismaciilsQ)
    -- Started music on hold, class 'hormuudmoh', on channel 'SIP/10.11.63.77-0000004b'
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 19270
Adding codec alaw to SDP
Adding codec opus to SDP
Adding codec ulaw to SDP
Adding codec g722 to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.12.35.175:58997:
INVITE sip:08bac4au@rouvg4b9ghhp.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.12.39.223:5060;branch=z9hG4bK577ae872
Max-Forwards: 70
From: "611693419" <sip:611693419@10.12.39.223>;tag=as365cc2ce
To: <sip:08bac4au@rouvg4b9ghhp.invalid;transport=ws>
Contact: <sip:611693419@10.12.39.223:5060;transport=ws>
Call-ID: 32afebed3f0e504819ca2afb5162a961@10.12.39.223:5060
CSeq: 102 INVITE
User-Agent: MyPBXServer9
Date: Tue, 03 Jan 2023 07:27:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "611693419" <sip:611693419@10.12.39.223>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 1030

v=0
o=root 215761295 215761295 IN IP4 10.12.39.223
s=Asterisk PBX 18.15.0
c=IN IP4 10.12.39.223
t=0 0
m=audio 19270 RTP/SAVPF 8 107 0 9 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=ice-ufrag:189e856b27ca4a773a6358fb7264fe8c
a=ice-pwd:61a0ed8f6e95eb1e3c03aa54406bc79f
a=candidate:Ha0c27df 1 UDP 2130706431 10.12.39.223 19270 typ host
a=candidate:Ha9fe5f78 1 UDP 2130706431 169.254.95.120 19270 typ host
a=candidate:Hac120001 1 UDP 2130706431 172.18.0.1 19270 typ host
a=candidate:Ha0c27df 2 UDP 2130706430 10.12.39.223 19271 typ host
a=candidate:Ha9fe5f78 2 UDP 2130706430 169.254.95.120 19271 typ host
a=candidate:Hac120001 2 UDP 2130706430 172.18.0.1 19271 typ host
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 FA:CD:5C:30:C3:0D:60:59:EB:3E:53:24:84:20:FE:20:23:08:6B:E8:7E:7C:DC:EF:83:45:FD:DE:4E:2B:4C:E7
a=rtcp-mux
a=sendrecv

---
    -- Called SIP/145356396110919750585317458531289112105
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
[Jan  3 10:27:49] ERROR[356612]: cdr.c:3357 ast_cdr_getvar: Unable to find CDR for channel SIP/666-0000004d
[Jan  3 10:27:49] ERROR[356612]: cdr.c:3357 ast_cdr_getvar: Unable to find CDR for channel SIP/666-0000004d
[Jan  3 10:27:49] ERROR[356612]: cdr.c:3357 ast_cdr_getvar: Unable to find CDR for channel SIP/666-0000004d

<--- SIP read from WS:10.12.35.175:58997 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS 10.12.39.223:5060;branch=z9hG4bK577ae872
To: <sip:08bac4au@rouvg4b9ghhp.invalid;transport=ws>
From: "611693419" <sip:611693419@10.12.39.223>;tag=as365cc2ce
Call-ID: 32afebed3f0e504819ca2afb5162a961@10.12.39.223:5060
CSeq: 102 INVITE
Supported: ice,replaces,outbound
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from WS:10.12.35.175:58997 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 10.12.39.223:5060;branch=z9hG4bK577ae872
To: <sip:08bac4au@rouvg4b9ghhp.invalid;transport=ws>;tag=7mo9orkjbv
From: "611693419" <sip:611693419@10.12.39.223>;tag=as365cc2ce
Call-ID: 32afebed3f0e504819ca2afb5162a961@10.12.39.223:5060
CSeq: 102 INVITE
Contact: <sip:08bac4au@rouvg4b9ghhp.invalid;transport=ws>
Supported: ice,replaces,outbound
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:08bac4au@rouvg4b9ghhp.invalid;transport=ws>
Audio is at 14268
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec ulaw to SDP
Adding codec ilbc to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.12.35.175:62981:
INVITE sip:666@10.12.35.175:62981;ob SIP/2.0
Via: SIP/2.0/UDP 10.12.39.223:5060;branch=z9hG4bK20b343e3
Max-Forwards: 70
From: "611693419" <sip:611693419@10.12.39.223>;tag=as653f5675
To: <sip:666@10.12.35.175:62981;ob>
Contact: <sip:611693419@10.12.39.223:5060>
Call-ID: 13c0c73e6047901a0e9418110b6aaa8d@10.12.39.223:5060
CSeq: 102 INVITE
User-Agent: MyPBXServer9
Date: Tue, 03 Jan 2023 07:27:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 344

v=0
o=root 1985408103 1985408103 IN IP4 10.12.39.223
s=Asterisk PBX 18.15.0
c=IN IP4 10.12.39.223
t=0 0
m=audio 14268 RTP/AVP 8 3 0 97 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
    -- Called SIP/666
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
[Jan  3 10:27:49] ERROR[356612]: cdr.c:3357 ast_cdr_getvar: Unable to find CDR for channel SIP/333-0000004e
[Jan  3 10:27:49] ERROR[356612]: cdr.c:3357 ast_cdr_getvar: Unable to find CDR for channel SIP/333-0000004e
[Jan  3 10:27:49] ERROR[356612]: cdr.c:3357 ast_cdr_getvar: Unable to find CDR for channel SIP/333-0000004e
Audio is at 19760
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec ulaw to SDP
Adding codec g722 to SDP
Adding codec opus to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.12.35.175:65161:
INVITE sip:bl4ut6kl@192.0.2.113;transport=wss SIP/2.0
Via: SIP/2.0/WS 10.12.39.223:5060;branch=z9hG4bK6366db8d;rport
Max-Forwards: 70
From: "611693419" <sip:611693419@10.12.39.223>;tag=as3520da5d
To: <sip:bl4ut6kl@192.0.2.113;transport=wss>
Contact: <sip:611693419@10.12.39.223:5060;transport=ws>
Call-ID: 2e7ff29567f742eb28c4685c4c31643f@10.12.39.223:5060
CSeq: 102 INVITE
User-Agent: MyPBXServer9
Date: Tue, 03 Jan 2023 07:27:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1030

v=0
o=root 310947332 310947332 IN IP4 10.12.39.223
s=Asterisk PBX 18.15.0
c=IN IP4 10.12.39.223
t=0 0
m=audio 19760 RTP/SAVPF 8 3 0 9 107 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=ice-ufrag:187930e25b2718700dfd9e832290ee7e
a=ice-pwd:0f4e304c52c05889354ee1e57c788c7a
a=candidate:Ha0c27df 1 UDP 2130706431 10.12.39.223 19760 typ host
a=candidate:Ha9fe5f78 1 UDP 2130706431 169.254.95.120 19760 typ host
a=candidate:Hac120001 1 UDP 2130706431 172.18.0.1 19760 typ host
a=candidate:Ha0c27df 2 UDP 2130706430 10.12.39.223 19761 typ host
a=candidate:Ha9fe5f78 2 UDP 2130706430 169.254.95.120 19761 typ host
a=candidate:Hac120001 2 UDP 2130706430 172.18.0.1 19761 typ host
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 FA:CD:5C:30:C3:0D:60:59:EB:3E:53:24:84:20:FE:20:23:08:6B:E8:7E:7C:DC:EF:83:45:FD:DE:4E:2B:4C:E7
a=rtcp-mux
a=sendrecv

---
    -- Called SIP/333
    -- SIP/145356396110919750585317458531289112105-0000004c connected line has changed. Saving it until answer for SIP/10.11.63.77-0000004b
    -- SIP/145356396110919750585317458531289112105-0000004c is ringing

<--- SIP read from UDP:10.12.35.175:62981 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.12.39.223:5060;received=10.12.39.223;branch=z9hG4bK20b343e3
Call-ID: 13c0c73e6047901a0e9418110b6aaa8d@10.12.39.223:5060
From: "611693419" <sip:611693419@10.12.39.223>;tag=as653f5675
To: <sip:666@10.12.35.175;ob>
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:10.12.35.175:62981 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.12.39.223:5060;received=10.12.39.223;branch=z9hG4bK20b343e3
Call-ID: 13c0c73e6047901a0e9418110b6aaa8d@10.12.39.223:5060
From: "611693419" <sip:611693419@10.12.39.223>;tag=as653f5675
To: <sip:666@10.12.35.175;ob>;tag=3575ffa3724740468887f300001d1be2
CSeq: 102 INVITE
Contact: <sip:666@10.12.35.175:62981;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:666@10.12.35.175:62981;ob>
    -- SIP/666-0000004d is ringing

<--- SIP read from WS:10.12.35.175:65161 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS 10.12.39.223:5060;branch=z9hG4bK6366db8d;rport
From: "611693419" <sip:611693419@10.12.39.223>;tag=as3520da5d
To: <sip:bl4ut6kl@192.0.2.113;transport=wss>
CSeq: 102 INVITE
Call-ID: 2e7ff29567f742eb28c4685c4c31643f@10.12.39.223:5060
Supported: outbound
User-Agent: Browser Phone 0.3.20 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/108.0.0.0 Safari/537.36
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from WS:10.12.35.175:65161 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 10.12.39.223:5060;branch=z9hG4bK6366db8d;rport
From: "611693419" <sip:611693419@10.12.39.223>;tag=as3520da5d
To: <sip:bl4ut6kl@192.0.2.113;transport=wss>;tag=29d63198gc
CSeq: 102 INVITE
Call-ID: 2e7ff29567f742eb28c4685c4c31643f@10.12.39.223:5060
Supported: outbound
User-Agent: Browser Phone 0.3.20 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/108.0.0.0 Safari/537.36
Contact: <sip:bl4ut6kl@192.0.2.113;transport=wss>
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:bl4ut6kl@192.0.2.113;transport=wss>
    -- SIP/333-0000004e is ringing
       > 0x7f0ba80639c0 -- Strict RTP learning complete - Locking on source address 10.11.5.200:4392
       > 0x7f0c1c0045d0 -- Strict RTP learning after remote address set to: 10.12.35.175:61629

<--- SIP read from WS:10.12.35.175:65161 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 10.12.39.223:5060;branch=z9hG4bK6366db8d;rport
From: "611693419" <sip:611693419@10.12.39.223>;tag=as3520da5d
To: <sip:bl4ut6kl@192.0.2.113;transport=wss>;tag=29d63198gc
CSeq: 102 INVITE
Call-ID: 2e7ff29567f742eb28c4685c4c31643f@10.12.39.223:5060
Supported: outbound
User-Agent: Browser Phone 0.3.20 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/108.0.0.0 Safari/537.36
Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,REGISTER,SUBSCRIBE
Contact: <sip:bl4ut6kl@192.0.2.113;transport=wss>
Content-Type: application/sdp
Content-Length: 1033

v=0
o=- 2585627047009063329 2 IN IP4 127.0.0.1
s=-
t=0 0
a=msid-semantic: WMS 8f682686-d5ed-4691-8931-a1e623af148a
m=audio 61629 RTP/SAVPF 8 0 9 107 101
c=IN IP4 192.168.20.32
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:3359826567 1 udp 2122260223 192.168.20.32 61629 typ host generation 0 network-id 1
a=candidate:2765909317 1 udp 2122194687 172.24.224.1 61630 typ host generation 0 network-id 2
a=ice-ufrag:+ql0
a=ice-pwd:jA0YjAOOcyd2JcZuk94EC9Re
a=ice-options:trickle
a=fingerprint:sha-256 58:C3:DE:A7:FA:68:13:38:22:F2:62:48:14:F3:E5:84:75:DB:B6:DB:71:8F:B7:71:44:59:B3:EA:25:AC:6A:92
a=setup:active
a=mid:0
a=sendrecv
a=msid:8f682686-d5ed-4691-8931-a1e623af148a e9b7bbea-c80e-4728-9714-6284e4467ff1
a=rtcp-mux
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 minptime=10;useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=ssrc:1864265884 cname:rfrxmkQrIjmqfXN6
a=ssrc:1864265884 msid:8f682686-d5ed-4691-8931-a1e623af148a e9b7bbea-c80e-4728-9714-6284e4467ff1
<------------->
--- (12 headers 27 lines) ---
Got SDP version 2 and unique parts [- 2585627047009063329 IN IP4 127.0.0.1]
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 9
Found RTP audio format 107
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G722 for ID 9
Found audio description format opus for ID 107
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g722|opus|vp9|vp8|h264), peer - audio=(ulaw|alaw|g722|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g722|opus)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.20.32:61629
sip_route_dump: route/path hop: <sip:bl4ut6kl@192.0.2.113;transport=wss>
Transmitting (NAT) to 10.12.35.175:65161:
ACK sip:bl4ut6kl@192.0.2.113;transport=wss SIP/2.0
Via: SIP/2.0/WS 10.12.39.223:5060;branch=z9hG4bK52b947ca;rport
Max-Forwards: 70
From: "611693419" <sip:611693419@10.12.39.223>;tag=as3520da5d
To: <sip:bl4ut6kl@192.0.2.113;transport=wss>;tag=29d63198gc
Contact: <sip:611693419@10.12.39.223:5060;transport=ws>
Call-ID: 2e7ff29567f742eb28c4685c4c31643f@10.12.39.223:5060
CSeq: 102 ACK
User-Agent: MyPBXServer9
Content-Length: 0


---
    -- SIP/333-0000004e answered SIP/10.11.63.77-0000004b
Scheduling destruction of SIP dialog '32afebed3f0e504819ca2afb5162a961@10.12.39.223:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 10.12.35.175:58997:
CANCEL sip:08bac4au@rouvg4b9ghhp.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.12.39.223:5060;branch=z9hG4bK577ae872
Max-Forwards: 70
From: "611693419" <sip:611693419@10.12.39.223>;tag=as365cc2ce
To: <sip:08bac4au@rouvg4b9ghhp.invalid;transport=ws>
Call-ID: 32afebed3f0e504819ca2afb5162a961@10.12.39.223:5060
CSeq: 102 CANCEL
User-Agent: MyPBXServer9
Reason: SIP;cause=200;text="Call completed elsewhere"
Content-Length: 0


---
Scheduling destruction of SIP dialog '32afebed3f0e504819ca2afb5162a961@10.12.39.223:5060' in 6400 ms (Method: INVITE)
Scheduling destruction of SIP dialog '13c0c73e6047901a0e9418110b6aaa8d@10.12.39.223:5060' in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 10.12.35.175:62981:
CANCEL sip:666@10.12.35.175:62981;ob SIP/2.0
Via: SIP/2.0/UDP 10.12.39.223:5060;branch=z9hG4bK20b343e3
Max-Forwards: 70
From: "611693419" <sip:611693419@10.12.39.223>;tag=as653f5675
To: <sip:666@10.12.35.175:62981;ob>
Call-ID: 13c0c73e6047901a0e9418110b6aaa8d@10.12.39.223:5060
CSeq: 102 CANCEL
User-Agent: MyPBXServer9
Reason: SIP;cause=200;text="Call completed elsewhere"
Content-Length: 0


---
Scheduling destruction of SIP dialog '13c0c73e6047901a0e9418110b6aaa8d@10.12.39.223:5060' in 32000 ms (Method: INVITE)
    -- Stopped music on hold on SIP/10.11.63.77-0000004b

<--- SIP read from WS:10.12.35.175:58997 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 10.12.39.223:5060;branch=z9hG4bK577ae872
To: <sip:08bac4au@rouvg4b9ghhp.invalid;transport=ws>;tag=bi8hnk3nod
From: "611693419" <sip:611693419@10.12.39.223>;tag=as365cc2ce
Call-ID: 32afebed3f0e504819ca2afb5162a961@10.12.39.223:5060
CSeq: 102 CANCEL
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from WS:10.12.35.175:58997 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/WS 10.12.39.223:5060;branch=z9hG4bK577ae872
To: <sip:08bac4au@rouvg4b9ghhp.invalid;transport=ws>;tag=7mo9orkjbv
From: "611693419" <sip:611693419@10.12.39.223>;tag=as365cc2ce
Call-ID: 32afebed3f0e504819ca2afb5162a961@10.12.39.223:5060
CSeq: 102 INVITE
Supported: ice,replaces,outbound
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 10.12.35.175:58997:
ACK sip:08bac4au@rouvg4b9ghhp.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.12.39.223:5060;branch=z9hG4bK577ae872
Max-Forwards: 70
From: "611693419" <sip:611693419@10.12.39.223>;tag=as365cc2ce
To: <sip:08bac4au@rouvg4b9ghhp.invalid;transport=ws>;tag=7mo9orkjbv
Contact: <sip:611693419@10.12.39.223:5060;transport=ws>
Call-ID: 32afebed3f0e504819ca2afb5162a961@10.12.39.223:5060
CSeq: 102 ACK
User-Agent: MyPBXServer9
Content-Length: 0


---
Scheduling destruction of SIP dialog '32afebed3f0e504819ca2afb5162a961@10.12.39.223:5060' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:10.12.35.175:62981 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.12.39.223:5060;received=10.12.39.223;branch=z9hG4bK20b343e3
Call-ID: 13c0c73e6047901a0e9418110b6aaa8d@10.12.39.223:5060
From: "611693419" <sip:611693419@10.12.39.223>;tag=as653f5675
To: <sip:666@10.12.35.175;ob>;tag=3575ffa3724740468887f300001d1be2
CSeq: 102 CANCEL
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:10.12.35.175:62981 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.12.39.223:5060;received=10.12.39.223;branch=z9hG4bK20b343e3
Call-ID: 13c0c73e6047901a0e9418110b6aaa8d@10.12.39.223:5060
From: "611693419" <sip:611693419@10.12.39.223>;tag=as653f5675
To: <sip:666@10.12.35.175;ob>;tag=3575ffa3724740468887f300001d1be2
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 10.12.35.175:62981:
ACK sip:666@10.12.35.175:62981;ob SIP/2.0
Via: SIP/2.0/UDP 10.12.39.223:5060;branch=z9hG4bK20b343e3
Max-Forwards: 70
From: "611693419" <sip:611693419@10.12.39.223>;tag=as653f5675
To: <sip:666@10.12.35.175:62981;ob>;tag=3575ffa3724740468887f300001d1be2
Contact: <sip:611693419@10.12.39.223:5060>
Call-ID: 13c0c73e6047901a0e9418110b6aaa8d@10.12.39.223:5060
CSeq: 102 ACK
User-Agent: MyPBXServer9
Content-Length: 0


---
Scheduling destruction of SIP dialog '13c0c73e6047901a0e9418110b6aaa8d@10.12.39.223:5060' in 32000 ms (Method: INVITE)
[Jan  3 10:28:00] WARNING[357082][C-00000019]: app.c:277 ast_app_exec_macro: Cannot run 'Macro(mymacro)'.  The application is not available.
    -- Channel SIP/333-0000004e joined 'simple_bridge' basic-bridge <c69114a1-1924-4af5-9c19-3722174b8ab4>
    -- Channel SIP/10.11.63.77-0000004b joined 'simple_bridge' basic-bridge <c69114a1-1924-4af5-9c19-3722174b8ab4>
       > 0x7f0c1c0045d0 -- Strict RTP learning after ICE completion
       > 0x7f0c1c0045d0 -- Strict RTP learning after remote address set to: 10.12.35.175:61629
       > 0x7f0c1c0045d0 -- Strict RTP switching to RTP target address 10.12.35.175:61629 as source
       > 0x7f0c1c0045d0 -- Strict RTP qualifying stream type: audio
       > 0x7f0c1c0045d0 -- Strict RTP switching source address to 192.168.20.32:61629
Really destroying SIP dialog 'gjmqomb0gdd7kgpov1h9d5' Method: REGISTER
       > 0x7f0c1c0045d0 -- Strict RTP learning complete - Locking on source address 192.168.20.32:61629
Really destroying SIP dialog '32afebed3f0e504819ca2afb5162a961@10.12.39.223:5060' Method: INVITE

<--- SIP read from UDP:10.11.63.77:5068 --->
BYE sip:2059@10.12.39.223:5060 SIP/2.0
Via: SIP/2.0/UDP 10.11.63.77:5068;branch=z9hG4bKnpiipfjiijxut7vwuipiou4xx;X-DispMsg=1407
Call-ID: ptwfiio47dtot7ofju4tf6fvnvdvvj4d@10.18.5.64
From: "611693419"<sip:611693419@10.11.63.77;transport=udp;user=phone>;tag=ivvxtjvv-CC-1004-OFC-71
To: "2059"<sip:2059@10.12.39.223;transport=udp;user=phone>;tag=as056f1cf4
CSeq: 3 BYE
Max-Forwards: 70
Reason: Q.850;cause=16;text="Normal call clearing"
Content-Length: 6
Content-Type: application/isup;version=itu-t92+

<------------->
--- (10 headers 1 lines) ---
Sending to 10.11.63.77:5068 (no NAT)
Scheduling destruction of SIP dialog 'ptwfiio47dtot7ofju4tf6fvnvdvvj4d@10.18.5.64' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 10.11.63.77:5068 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.11.63.77:5068;branch=z9hG4bKnpiipfjiijxut7vwuipiou4xx;X-DispMsg=1407;received=10.11.63.77
From: "611693419"<sip:611693419@10.11.63.77;transport=udp;user=phone>;tag=ivvxtjvv-CC-1004-OFC-71
To: "2059"<sip:2059@10.12.39.223;transport=udp;user=phone>;tag=as056f1cf4
Call-ID: ptwfiio47dtot7ofju4tf6fvnvdvvj4d@10.18.5.64
CSeq: 3 BYE
Server: MyPBXServer9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
    -- Channel SIP/10.11.63.77-0000004b left 'simple_bridge' basic-bridge <c69114a1-1924-4af5-9c19-3722174b8ab4>
    -- Channel SIP/333-0000004e left 'simple_bridge' basic-bridge <c69114a1-1924-4af5-9c19-3722174b8ab4>
Scheduling destruction of SIP dialog '2e7ff29567f742eb28c4685c4c31643f@10.12.39.223:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 10.12.35.175:65161:
BYE sip:bl4ut6kl@192.0.2.113;transport=wss SIP/2.0
Via: SIP/2.0/WS 10.12.39.223:5060;branch=z9hG4bK7b7f8aa2;rport
Max-Forwards: 70
From: "611693419" <sip:611693419@10.12.39.223>;tag=as3520da5d
To: <sip:bl4ut6kl@192.0.2.113;transport=wss>;tag=29d63198gc
Call-ID: 2e7ff29567f742eb28c4685c4c31643f@10.12.39.223:5060
CSeq: 103 BYE
User-Agent: MyPBXServer9
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from WS:10.12.35.175:65161 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 10.12.39.223:5060;branch=z9hG4bK7b7f8aa2;rport
From: "611693419" <sip:611693419@10.12.39.223>;tag=as3520da5d
To: <sip:bl4ut6kl@192.0.2.113;transport=wss>;tag=29d63198gc
CSeq: 103 BYE
Call-ID: 2e7ff29567f742eb28c4685c4c31643f@10.12.39.223:5060
Supported: outbound
User-Agent: Browser Phone 0.3.20 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/108.0.0.0 Safari/537.36
Content-Length: 0

<------------->

david551 · January 3, 2023, 11:51am

Please mark up your configuration and logs as preformatted text, as otherwise the forum garbles them. E.g. the above extract is missing the templates, that are only visible in the the raw version, at the moment:

[333](basic,webrtc)
username=333
secret=mysecrate

ahmedna · January 3, 2023, 4:05pm

@david551 thank you for your notice i did this was my first time
do you have idea solving one way audio in webRTC

system · February 2, 2023, 4:06pm

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.