Missing sdp rtpmap for dyanmic payload type erro when MIZUDROID android app to call to webrtc browser

I create on angular application basically it is used Webrtc and second end I install one application for MIZUDROID in android application my problem is that when I call from chrome borwser using angular applcation call is going on and perfectly I can hear sound from browser to Andorid appp but when I call from MIZUDROID app to angular app at that time call is not coming and eroor say
“missing sdp rtpmap for dyanmic payload type”
Any one suggest how I can remove this solved. I have already public ip and here my pjsip.conf, extension.conf,rtp.conf,http.conf and modules.conf what setting I missed please suggest

  1. pjsip.conf
    =============
    [transport-wss]
    type=transport
    protocol=wss
    bind=0.0.0.0

[199]
type=endpoint
aors=199
auth=199
use_avpf=yes
media_encryption=dtls
dtls_ca_file=/etc/asterisk/keys/asterisk.pem
dtls_cert_file=/etc/asterisk/keys/asterisk.pem
dtls_verify=fingerprint
dtls_setup=actpass
ice_support=yes
media_use_received_transport=yes
rtcp_mux=yes
context=helloworld2
disallow=all
allow=alaw,ulaw
allow=opus
;device=hw:0,0
;direct_media=no
;rtp_symmetric=yes
;force_rport=yes
;rewrite_contact=yes
;external_media_address=35.244.39.248
;external_signaling_address=35.244.39.248

[199]
type=auth
auth_type=userpass
username=199
password=test

[199]
type=aor
max_contacts=1
remove_existing=yes

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0

[3001]
type=endpoint
context=helloworld
disallow=all
allow=ulaw
auth=3001
aors=3001

[3001]
type=auth
auth_type=userpass
password=3001pass
username=3001

[3001]
type=aor
max_contacts=1
remove_existing=yes

[3002]
type=endpoint
context=helloworld2
disallow=all
allow=ulaw
auth=3002
aors=3002

[3002]
type=auth
auth_type=userpass
password=3002pass
username=3002

[3002]
type=aor
max_contacts=1
remove_existing=yes

  1. extensions.conf
    =================
    [helloworld]
    exten => _X.,1,NoOp(${EXTEN})
    same => n,Playback(hello-world)
    same => n,Hangup()
    [helloworld2]
    exten => _X.,1,NoOp(${EXTEN})
    same => n,Playback(hello-world)
    same => n,Dial(PJSIP/${EXTEN},20)
    same => n,Read(Digits,)
    same => n,Playback(you-entered)
    same => n,SayNumber(${Digits})

  2. rtp.conf
    ============
    [general]
    rtpstart=10000
    rtpend=20000
    icesupport=true
    stunaddr:19302=stun.l.google.com
    4.http.conf
    [general]
    enabled=yes
    bindaddr=0.0.0.0
    bindport=8088
    tlsenable=yes
    tlsbindaddr=0.0.0.0:8089
    tlscertfile=/etc/asterisk/keys/asterisk.pem

  3. modules.conf
    ======================
    ;
    ; Asterisk configuration file
    ;
    ; Module Loader configuration file
    ;
    [modules]
    autoload=yes
    ;
    ; Any modules that need to be loaded before the Asterisk core has been
    ; initialized (just after the logger has been initialized) can be loaded
    ; using ‘preload’. This will frequently be needed if you wish to map all
    ; module configuration files into Realtime storage, since the Realtime
    ; driver will need to be loaded before the modules using those configuration
    ; files are initialized.
    ;
    ; An example of loading ODBC support would be:
    ;preload => res_odbc.so
    ;preload => res_config_odbc.so
    ;
    ; If you want, load the GTK console right away.
    ; Don’t load the KDE console since
    ; it’s not as sophisticated right now.
    ;
    noload => pbx_gtkconsole.so
    ;load => pbx_gtkconsole.so
    noload => pbx_kdeconsole.so
    ;
    ; Intercom application is obsoleted by
    ; chan_oss. Don’t load it.
    ;
    noload => app_intercom.so
    ;
    ; The ‘modem’ channel driver and its subdrivers are
    ; obsolete, don’t load them.
    ;
    noload => chan_modem.so
    noload => chan_modem_aopen.so
    noload => chan_modem_bestdata.so
    noload => chan_modem_i4l.so
    ;
    ; Comment this out (after installing CAPI middleware and hardware
    ; drivers) if you have CAPI-able hardware and wish to use it in
    ; Asterisk.
    ;
    noload => chan_capi.so
    ;
    load => res_musiconhold.so
    ;
    ; Do not load load local channel drivers (using the system speaker) by default,
    ; they are not used in most installations and might block the sound hardware
    ;
    noload => chan_alsa.so
    noload => chan_console.so
    noload => chan_oss.so
    ;
    ; Disable CDR logging to SQLite by default since it writes unconditionally to
    ; cdr.db without a way to rotate it.
    ;
    noload => cdr_sqlite.so
    ;
    ; These conflict with app_directory.so and each other.
    noload => app_directory_odbc.so
    ;
    ; Enable these if you want to configure Asterisk in a database
    ;
    noload => res_config_odbc.so
    noload => res_config_pgsql.so
    ;
    ; Module names listed in “global” section will have symbols globally
    ; exported to modules loaded after them.
    ;
    ;[global]
    noload => chan_sip.so

I have already public ip purchased and my private ip is 10.111.2.21

Please suggest any way for solved this issue. and I am using Asterisk 18.10.0 this version.

You haven’t provided a SIP trace using “pjsip set logger on” which would show what the device is offering. Based on the limited information, though, I would assume a bug or issue with the Android app.

As per you suggestion I trace the app it show following logs

<— Transmitting SIP response (793 bytes) to UDP:27.61.153.115:10130 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 27.61.153.115:10130;rport=10130;received=27.61.153.115;branch=z9hG4bK.oA9-ppkae
Call-ID: vBgOAQEb3k
From: sip:3001@35.244.39.248;tag=32AHBcB7z
To: sip:199@35.244.39.248;tag=01d3d599-4195-47d7-9737-cd8f8116aafe
CSeq: 21 INVITE
Server: Asterisk PBX 18.10.0~dfsg+~cs6.10.40431411-2
Contact: sip:10.111.2.21:5060
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 223

v=0
o=- 3525 1303 IN IP4 10.111.2.21
s=Asterisk
c=IN IP4 10.111.2.21
t=0 0
m=audio 18856 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Transmitting SIP response (793 bytes) to UDP:27.61.153.115:10130 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 27.61.153.115:10130;rport=10130;received=27.61.153.115;branch=z9hG4bK.oA9-ppkae
Call-ID: vBgOAQEb3k
From: sip:3001@35.244.39.248;tag=32AHBcB7z
To: sip:199@35.244.39.248;tag=01d3d599-4195-47d7-9737-cd8f8116aafe
CSeq: 21 INVITE
Server: Asterisk PBX 18.10.0~dfsg+~cs6.10.40431411-2
Contact: sip:10.111.2.21:5060
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 223

v=0
o=- 3525 1303 IN IP4 10.111.2.21
s=Asterisk
c=IN IP4 10.111.2.21
t=0 0
m=audio 18856 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Transmitting SIP request (401 bytes) to UDP:27.61.153.115:33149 —>
BYE sip:3001@27.61.153.115:33149;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.111.2.21:5060;rport;branch=z9hG4bKPj538d962b-b631-44a5-a888-b3e87e6c7b24
From: sip:199@35.244.39.248;tag=01d3d599-4195-47d7-9737-cd8f8116aafe
To: sip:3001@35.244.39.248;tag=32AHBcB7z
Call-ID: vBgOAQEb3k
CSeq: 4806 BYE
Max-Forwards: 70
User-Agent: Asterisk PBX 18.10.0~dfsg+~cs6.10.40431411-2
Content-Length: 0

<— Transmitting SIP request (401 bytes) to UDP:27.61.153.115:33149 —>
BYE sip:3001@27.61.153.115:33149;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.111.2.21:5060;rport;branch=z9hG4bKPj538d962b-b631-44a5-a888-b3e87e6c7b24
From: sip:199@35.244.39.248;tag=01d3d599-4195-47d7-9737-cd8f8116aafe
To: sip:3001@35.244.39.248;tag=32AHBcB7z
Call-ID: vBgOAQEb3k
CSeq: 4806 BYE
Max-Forwards: 70
User-Agent: Asterisk PBX 18.10.0~dfsg+~cs6.10.40431411-2
Content-Length: 0

<— Transmitting SIP request (401 bytes) to UDP:27.61.153.115:33149 —>
BYE sip:3001@27.61.153.115:33149;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.111.2.21:5060;rport;branch=z9hG4bKPj538d962b-b631-44a5-a888-b3e87e6c7b24
From: sip:199@35.244.39.248;tag=01d3d599-4195-47d7-9737-cd8f8116aafe
To: sip:3001@35.244.39.248;tag=32AHBcB7z
Call-ID: vBgOAQEb3k
CSeq: 4806 BYE
Max-Forwards: 70
User-Agent: Asterisk PBX 18.10.0~dfsg+~cs6.10.40431411-2
Content-Length: 0

<— Received SIP request (423 bytes) from UDP:94.23.152.160:54919 —>
REGISTER sip:35.244.39.248 SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0:54919;branch=z9hG4bK1220490055
Max-Forwards: 70
From: sip:9084@35.244.39.248;tag=824161036
To: sip:9084@35.244.39.248
Call-ID: 24933606-583375149-2093618376
CSeq: 1 REGISTER
Contact: sip:9084@192.168.1.3:54919
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: pplsip
Content-Length: 0

What It mean I can not uderstand sir

That is incomplete and doesn’t show a complete attempt.

Call is conduct but no sound from both side any suggestion please

<— Transmitting SIP response (793 bytes) to UDP:27.61.153.115:10134 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 27.61.153.115:10134;rport=10134;received=27.61.153.115;branch=z9hG4bK.w47hRlprM
Call-ID: et8NjCF2ST
From: sip:3001@35.244.39.248;tag=VXTs7SY~m
To: sip:199@35.244.39.248;tag=6ad94bd9-1c2e-4cb9-aef0-7306a6cde568
CSeq: 21 INVITE
Server: Asterisk PBX 18.10.0~dfsg+~cs6.10.40431411-2
Contact: sip:10.111.2.21:5060
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 223

v=0
o=- 2714 2221 IN IP4 10.111.2.21
s=Asterisk
c=IN IP4 10.111.2.21
t=0 0
m=audio 13414 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Transmitting SIP response (793 bytes) to UDP:27.61.153.115:10134 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 27.61.153.115:10134;rport=10134;received=27.61.153.115;branch=z9hG4bK.w47hRlprM
Call-ID: et8NjCF2ST
From: sip:3001@35.244.39.248;tag=VXTs7SY~m
To: sip:199@35.244.39.248;tag=6ad94bd9-1c2e-4cb9-aef0-7306a6cde568
CSeq: 21 INVITE
Server: Asterisk PBX 18.10.0~dfsg+~cs6.10.40431411-2
Contact: sip:10.111.2.21:5060
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 223

v=0
o=- 2714 2221 IN IP4 10.111.2.21
s=Asterisk
c=IN IP4 10.111.2.21
t=0 0
m=audio 13414 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Transmitting SIP response (793 bytes) to UDP:27.61.153.115:10134 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 27.61.153.115:10134;rport=10134;received=27.61.153.115;branch=z9hG4bK.w47hRlprM
Call-ID: et8NjCF2ST
From: sip:3001@35.244.39.248;tag=VXTs7SY~m
To: sip:199@35.244.39.248;tag=6ad94bd9-1c2e-4cb9-aef0-7306a6cde568
CSeq: 21 INVITE
Server: Asterisk PBX 18.10.0~dfsg+~cs6.10.40431411-2
Contact: sip:10.111.2.21:5060
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 223

v=0
o=- 2714 2221 IN IP4 10.111.2.21
s=Asterisk
c=IN IP4 10.111.2.21
t=0 0
m=audio 13414 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

this is from all logs
<— Transmitting SIP response (793 bytes) to UDP:27.61.153.115:10134 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 27.61.153.115:10134;rport=10134;received=27.61.153.115;branch=z9hG4bK.w47hRlprM
Call-ID: et8NjCF2ST
From: sip:3001@35.244.39.248;tag=VXTs7SY~m
To: sip:199@35.244.39.248;tag=6ad94bd9-1c2e-4cb9-aef0-7306a6cde568
CSeq: 21 INVITE
Server: Asterisk PBX 18.10.0~dfsg+~cs6.10.40431411-2
Contact: sip:10.111.2.21:5060
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 223

v=0
o=- 2714 2221 IN IP4 10.111.2.21
s=Asterisk
c=IN IP4 10.111.2.21
t=0 0
m=audio 13414 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Transmitting SIP response (793 bytes) to UDP:27.61.153.115:10134 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 27.61.153.115:10134;rport=10134;received=27.61.153.115;branch=z9hG4bK.w47hRlprM
Call-ID: et8NjCF2ST
From: sip:3001@35.244.39.248;tag=VXTs7SY~m
To: sip:199@35.244.39.248;tag=6ad94bd9-1c2e-4cb9-aef0-7306a6cde568
CSeq: 21 INVITE
Server: Asterisk PBX 18.10.0~dfsg+~cs6.10.40431411-2
Contact: sip:10.111.2.21:5060
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 223

v=0
o=- 2714 2221 IN IP4 10.111.2.21
s=Asterisk
c=IN IP4 10.111.2.21
t=0 0
m=audio 13414 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Transmitting SIP request (402 bytes) to UDP:27.61.153.115:43923 —>
BYE sip:3001@27.61.153.115:43923;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.111.2.21:5060;rport;branch=z9hG4bKPjdb4beabf-4e04-4e6d-bcd7-74baaf0b6ed6
From: sip:199@35.244.39.248;tag=6ad94bd9-1c2e-4cb9-aef0-7306a6cde568
To: sip:3001@35.244.39.248;tag=VXTs7SY~m
Call-ID: et8NjCF2ST
CSeq: 13846 BYE
Max-Forwards: 70
User-Agent: Asterisk PBX 18.10.0~dfsg+~cs6.10.40431411-2
Content-Length: 0

== WebSocket connection from ‘27.109.4.186:52632’ for protocol ‘sip’ accepted using version ‘13’
<— Transmitting SIP request (402 bytes) to UDP:27.61.153.115:43923 —>
BYE sip:3001@27.61.153.115:43923;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.111.2.21:5060;rport;branch=z9hG4bKPjdb4beabf-4e04-4e6d-bcd7-74baaf0b6ed6
From: sip:199@35.244.39.248;tag=6ad94bd9-1c2e-4cb9-aef0-7306a6cde568
To: sip:3001@35.244.39.248;tag=VXTs7SY~m
Call-ID: et8NjCF2ST
CSeq: 13846 BYE
Max-Forwards: 70
User-Agent: Asterisk PBX 18.10.0~dfsg+~cs6.10.40431411-2
Content-Length: 0

You have not configured PJSIP to know it is behind NAT. This is done using external_signaling_address, external_media_address, and local_net on the transport.

Sir Please suggest how to configrue please in pjsip. which place with which parameter.

Where I configure

In the pjsip.conf file on the respective transport.

[transport-wss]
type=transport
protocol=wss
bind=0.0.0.0
local_net=10.111.2.21
disallow=all
allow=opus,ulaw,alaw,vp8,h264
external_media_address=35.244.39.248
external_signaling_address=35.244.39.248
it is already in my pjsip.conf

That’s a websocket transport. The softphone is using UDP, and thus that transport would not apply.

Now what I do for call the browser? Please suggest

any changes needed for this?

What? You said previously the browser was working. If there’s some other problem then you need to state it, or state what is now going on. We don’t know what is going on beyond what you state here.

Browser was working in which case when I call from browser to android app. But not working when android app to browser.
Only one way working fine that is browser to phone.
Not working in case phone to browser.

So, did you actually configure Asterisk to know it is behind NAT, restart Asterisk, and test again? If so then you need to provide an updated SIP trace from the start of the call to the end of the call. Not just a portion of it, otherwise this is going to be going around in circles multiple times. You also need to ensure the RTP ports are forwarded as well.

Your problem, based on the limited information given, is not phone to browser. It is phone to Asterisk.