After receiving call no Sound coming at either End

Hello,
I have successfully registered SIP client at Asterisk Server.
But when I am trying to call the Mobile Linphone is ringing. But Not received any Voice at either End.
The below is the log I have received So far… xx.xx.xx.xx is my Asterisk Server address hosted at cloud and It is a Public IP address.

<--- SIP read from UDP:103.77.138.124:6666 --->
REGISTER sip:xx.xx.xx.xx:5060;maddr=xx.xx.xx.xx SIP/2.0
Via: SIP/2.0/UDP 10.28.113.30:5060;branch=z9hG4bK-3633-ffabbbb3dcd1705a1567754ddb9af3ce
Max-Forwards: 70
From: <sip:David@xx.xx.xx.xx>;tag=textClient
To: <sip:David@xx.xx.xx.xx>
Call-ID: f73caf64144d2cc1d65a75f7648c3d5d@10.28.113.30
CSeq: 2 REGISTER
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Contact: <sip:David@10.28.113.30:6666;transport=udp>
Expires: 3060
Authorization: Digest username="David",realm="asterisk",nonce="7b4da517",uri="sip:xx.xx.xx.xx:5060;maddr=xx.xx.xx.xx",response="1541c6dbcb33e0f65e0e5fabd496b1ae",algorithm=MD5
Content-Length: 0
=============
--- (12 headers 0 lines) ---
Sending to 103.77.138.124:6666 (NAT)
Reliably Transmitting (NAT) to 103.77.138.124:6666:
OPTIONS sip:David@10.28.113.30:6666;transport=udp SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK18781504;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@xx.xx.xx.xx>;tag=as78e5b4da
To: <sip:David@10.28.113.30:6666;transport=udp>
Contact: <sip:asterisk@xx.xx.xx.xx:5060>
Call-ID: 54a7b1f320190cbb52fa370b0c401023@xx.xx.xx.xx:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.7.0
Date: Fri, 24 Jul 2020 08:26:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
============<--- Transmitting (NAT) to 103.77.138.124:6666 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.28.113.30:5060;branch=z9hG4bK-3633-ffabbbb3dcd1705a1567754ddb9af3ce;received=103.77.138.124;rport=6666
From: <sip:David@xx.xx.xx.xx>;tag=textClient
To: <sip:David@xx.xx.xx.xx>;tag=as37e6c212
Call-ID: f73caf64144d2cc1d65a75f7648c3d5d@10.28.113.30
CSeq: 2 REGISTER
Server: Asterisk PBX 16.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 3060
Contact: <sip:David@10.28.113.30:6666;transport=udp>;expires=3060
Date: Fri, 24 Jul 2020 08:26:36 GMT
Content-Length: 0
===========<------------>
Scheduling destruction of SIP dialog 'f73caf64144d2cc1d65a75f7648c3d5d@10.28.113.30' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:103.77.138.124:6666 --->
SIP/2.0 200 OK
CSeq: 102 OPTIONS
Call-ID: 54a7b1f320190cbb52fa370b0c401023@xx.xx.xx.xx:5060
From: "asterisk" <sip:asterisk@xx.xx.xx.xx>;tag=as78e5b4da
To: <sip:David@10.28.113.30:6666;transport=udp>
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK18781504;rport=5060;received=xx.xx.xx.xx
Contact: <sip:David@10.28.113.30:6666;transport=udp>
Content-Length: 0
===============
--- (8 headers 0 lines) ---
Really destroying SIP dialog '54a7b1f320190cbb52fa370b0c401023@xx.xx.xx.xx:5060' Method: OPTIONS

<--- SIP read from UDP:103.77.138.124:6666 --->
INVITE sip:852@xx.xx.xx.xx SIP/2.0
CSeq: 2 INVITE
To: <sip:Smith@xx.xx.xx.xx>
Call-ID: 9b37308a55a2fac98d8bc4864b7fd38f@10.28.113.30
From: <sip:David@xx.xx.xx.xx>;tag=textClient
Max-Forwards: 70
Contact: <sip:David@10.28.113.30:6666;transport=udp>
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-3633-d578df4ffa41c2ae90722d87b35f9a09
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 103.77.138.124:6666 (NAT)
Sending to 103.77.138.124:6666 (NAT)
Using INVITE request as basis request - 9b37308a55a2fac98d8bc4864b7fd38f@10.28.113.30
Found peer 'David' for 'David' from 103.77.138.124:6666

<--- Reliably Transmitting (NAT) to 103.77.138.124:6666 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-3633-d578df4ffa41c2ae90722d87b35f9a09;received=103.77.138.124;rport=6666
From: <sip:David@xx.xx.xx.xx>;tag=textClient
To: <sip:Smith@xx.xx.xx.xx>;tag=as53b9c771
Call-ID: 9b37308a55a2fac98d8bc4864b7fd38f@10.28.113.30
CSeq: 2 INVITE
Server: Asterisk PBX 16.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="606625aa"
Content-Length: 0


<------------>
<--- SIP read from UDP:103.77.138.124:6666 --->
ACK sip:852@xx.xx.xx.xx SIP/2.0
Call-ID: 9b37308a55a2fac98d8bc4864b7fd38f@10.28.113.30
Max-Forwards: 70
From: <sip:David@xx.xx.xx.xx>;tag=textClient
To: <sip:Smith@xx.xx.xx.xx>;tag=as53b9c771
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-3633-d578df4ffa41c2ae90722d87b35f9a09
CSeq: 2 ACK
Content-Length: 0

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

<--- SIP read from UDP:103.77.138.124:6666 --->
INVITE sip:852@xx.xx.xx.xx:5060;maddr=xx.xx.xx.xx SIP/2.0
CSeq: 3 INVITE
To: <sip:Smith@xx.xx.xx.xx>
Call-ID: 9b37308a55a2fac98d8bc4864b7fd38f@10.28.113.30
From: <sip:David@xx.xx.xx.xx>;tag=textClient
Max-Forwards: 70
Contact: <sip:David@10.28.113.30:6666;transport=udp>
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-3633-72c2c45d0aea33e399bf65e58455a805
Authorization: Digest username="David",realm="asterisk",nonce="606625aa",uri="sip:852@xx.xx.xx.xx:5060;maddr=xx.xx.xx.xx",response="104f273ac12d689c96c5d29e4a87aacc",algorithm=MD5
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 103.77.138.124:6666 (NAT)
Sending to 103.77.138.124:6666 (NAT)
Using INVITE request as basis request - 9b37308a55a2fac98d8bc4864b7fd38f@10.28.113.30
Found peer 'David' for 'David' from 103.77.138.124:6666

<--- Reliably Transmitting (NAT) to 103.77.138.124:6666 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-3633-72c2c45d0aea33e399bf65e58455a805;received=103.77.138.124;rport=6666
From: <sip:David@xx.xx.xx.xx>;tag=textClient
To: <sip:Smith@xx.xx.xx.xx>;tag=as7e2c5123
Call-ID: 9b37308a55a2fac98d8bc4864b7fd38f@10.28.113.30
CSeq: 3 INVITE
Server: Asterisk PBX 16.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="396b0012"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '9b37308a55a2fac98d8bc4864b7fd38f@10.28.113.30' in 11840 ms (Method: INVITE)

<--- SIP read from UDP:103.77.138.124:6666 --->
ACK sip:852@xx.xx.xx.xx:5060;maddr=xx.xx.xx.xx SIP/2.0
Call-ID: 9b37308a55a2fac98d8bc4864b7fd38f@10.28.113.30
Max-Forwards: 70
From: <sip:David@xx.xx.xx.xx>;tag=textClient
To: <sip:Smith@xx.xx.xx.xx>;tag=as7e2c5123
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-3633-72c2c45d0aea33e399bf65e58455a805
CSeq: 3 ACK
Content-Length: 0

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

<--- SIP read from UDP:103.77.138.124:6666 --->
INVITE sip:852@xx.xx.xx.xx:5060;maddr=xx.xx.xx.xx SIP/2.0
CSeq: 4 INVITE
To: <sip:Smith@xx.xx.xx.xx>
Call-ID: 9b37308a55a2fac98d8bc4864b7fd38f@10.28.113.30
From: <sip:David@xx.xx.xx.xx>;tag=textClient
Max-Forwards: 70
Contact: <sip:David@10.28.113.30:6666;transport=udp>
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-3633-27af22b3fa8adb8d033b120ea62b1c43
Authorization: Digest username="David",realm="asterisk",nonce="396b0012",uri="sip:852@xx.xx.xx.xx:5060;maddr=xx.xx.xx.xx",response="83233e7dd0ebae10fcf1f4facdde8cb2",algorithm=MD5
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 103.77.138.124:6666 (NAT)
Using INVITE request as basis request - 9b37308a55a2fac98d8bc4864b7fd38f@10.28.113.30
Found peer 'David' for 'David' from 103.77.138.124:6666
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
Looking for 852 in incoming (domain xx.xx.xx.xx)
sip_route_dump: route/path hop: <sip:David@10.28.113.30:6666;transport=udp>

<--- Transmitting (NAT) to 103.77.138.124:6666 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-3633-27af22b3fa8adb8d033b120ea62b1c43;received=103.77.138.124;rport=6666
From: <sip:David@xx.xx.xx.xx>;tag=textClient
To: <sip:Smith@xx.xx.xx.xx>
Call-ID: 9b37308a55a2fac98d8bc4864b7fd38f@10.28.113.30
CSeq: 4 INVITE
Server: Asterisk PBX 16.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:852@xx.xx.xx.xx:5060>
Content-Length: 0


<------------>
    -- Executing [852@incoming:1] Dial("SIP/David-000000dd", "SIP/Smith")
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
Audio is at 13858
Video is at xx.xx.xx.xx:10322
Adding codec ulaw to SDP
Adding video codec h263 to SDP
Adding video codec h263p to SDP
Adding video codec vp8 to SDP
Adding video codec vp9 to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 103.77.138.124:54622:
INVITE sip:Smith@103.77.138.124:54622;transport=udp SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK757a12ba;rport
Max-Forwards: 70
From: <sip:David@xx.xx.xx.xx>;tag=as24fa164f
To: <sip:Smith@103.77.138.124:54622;transport=udp>
Contact: <sip:David@xx.xx.xx.xx:5060>
Call-ID: 582afea40f2f16120965162074049483@xx.xx.xx.xx:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.7.0
Date: Fri, 24 Jul 2020 08:26:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 446

v=0
o=root 826047459 826047459 IN IP4 xx.xx.xx.xx
s=Asterisk PBX 16.7.0
c=IN IP4 xx.xx.xx.xx
b=CT:384
t=0 0
m=audio 13858 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 10322 RTP/AVP 34 103 100 108
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=rtpmap:100 VP8/90000
a=rtcp-fb:* ccm fir
a=rtpmap:108 VP9/90000
a=sendrecv

---
---
  == Connect attempt from '195.154.31.140' unable to authenticate
Retransmitting #1 (NAT) to 103.77.138.124:54622:
INVITE sip:Smith@103.77.138.124:54622;transport=udp SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK757a12ba;rport
Max-Forwards: 70
From: <sip:David@xx.xx.xx.xx>;tag=as24fa164f
To: <sip:Smith@103.77.138.124:54622;transport=udp>
Contact: <sip:David@xx.xx.xx.xx:5060>
Call-ID: 582afea40f2f16120965162074049483@xx.xx.xx.xx:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.7.0
Date: Fri, 24 Jul 2020 08:26:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 446

v=0
o=root 826047459 826047459 IN IP4 xx.xx.xx.xx
s=Asterisk PBX 16.7.0
c=IN IP4 xx.xx.xx.xx
b=CT:384
t=0 0
m=audio 13858 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 10322 RTP/AVP 34 103 100 108
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=rtpmap:100 VP8/90000
a=rtcp-fb:* ccm fir
a=rtpmap:108 VP9/90000
a=sendrecv

---

<--- SIP read from UDP:103.77.138.124:54622 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK757a12ba;rport
From: <sip:David@xx.xx.xx.xx>;tag=as24fa164f
To: <sip:Smith@103.77.138.124:54622;transport=udp>
Call-ID: 582afea40f2f16120965162074049483@xx.xx.xx.xx:5060
CSeq: 102 INVITE

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

<--- SIP read from UDP:103.77.138.124:54622 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK757a12ba;rport
From: <sip:David@xx.xx.xx.xx>;tag=as24fa164f
To: <sip:Smith@103.77.138.124:54622;transport=udp>;tag=P1vhbK~
Call-ID: 582afea40f2f16120965162074049483@xx.xx.xx.xx:5060
CSeq: 102 INVITE
User-Agent: LinphoneAndroid/4.3.0 (Galaxy M30s) LinphoneSDK/4.4.0 (tags/4.4.0^0)
Supported: replaces, outbound, gruu

<------------->
--- (8 headers 0 lines) ---
sip_route_dump: no route/path
    -- SIP/Smith-000000de is ringing

<--- Transmitting (NAT) to 103.77.138.124:6666 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-3633-27af22b3fa8adb8d033b120ea62b1c43;received=103.77.138.124;rport=6666
From: <sip:David@xx.xx.xx.xx>;tag=textClient
To: <sip:Smith@xx.xx.xx.xx>;tag=as5340f7c4
Call-ID: 9b37308a55a2fac98d8bc4864b7fd38f@10.28.113.30
CSeq: 4 INVITE
Server: Asterisk PBX 16.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:852@xx.xx.xx.xx:5060>
Content-Length: 0


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

<--- SIP read from UDP:103.77.138.124:54622 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK757a12ba;rport
From: <sip:David@xx.xx.xx.xx>;tag=as24fa164f
To: <sip:Smith@103.77.138.124:54622;transport=udp>;tag=P1vhbK~
Call-ID: 582afea40f2f16120965162074049483@xx.xx.xx.xx:5060
CSeq: 102 INVITE
User-Agent: LinphoneAndroid/4.3.0 (Galaxy M30s) LinphoneSDK/4.4.0 (tags/4.4.0^0)
Supported: replaces, outbound, gruu

<------------->
--- (8 headers 0 lines) ---
sip_route_dump: no route/path
    -- SIP/Smith-000000de is ringing

Now the below is the RTP log

<--- SIP read from UDP:103.77.138.124:54622 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK427e6c57;rport
From: <sip:David@xx.xx.xx.xx>;tag=as4e0ef393
To: <sip:Smith@103.77.138.124:54622;transport=udp>;tag=UcVXKNG
Call-ID: 0e58956c4abddec0513ccd8c3fdb0d33@xx.xx.xx.xx:5060
CSeq: 102 INVITE
User-Agent: LinphoneAndroid/4.3.0 (Galaxy M30s) LinphoneSDK/4.4.0 (tags/4.4.0^0)
Supported: replaces, outbound, gruu
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Contact: <sip:Smith@103.77.138.124:54622;transport=udp>;expires=3600;+sip.instance="<urn:uuid:e9524773-5da2-00db-bc7d-9347a3a628a2>"
Content-Type: application/sdp
Content-Length: 180

v=0
o=Smith 200 1410 IN IP4 10.28.113.30
s=Talk
c=IN IP4 10.28.113.30
t=0 0
m=audio 7078 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
m=video 0 RTP/AVP 0
a=inactive
<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|h263|h263p|vp8|vp9), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.28.113.30:7078
Peer doesn't provide video
sip_route_dump: route/path hop: <sip:Smith@103.77.138.124:54622;transport=udp>
Transmitting (NAT) to 103.77.138.124:54622:
ACK sip:Smith@103.77.138.124:54622;transport=udp SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK69163c31;rport
Max-Forwards: 70
From: <sip:David@xx.xx.xx.xx>;tag=as4e0ef393
To: <sip:Smith@103.77.138.124:54622;transport=udp>;tag=UcVXKNG
Contact: <sip:David@xx.xx.xx.xx:5060>
Call-ID: 0e58956c4abddec0513ccd8c3fdb0d33@xx.xx.xx.xx:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.7.0
Content-Length: 0


---
    -- SIP/Smith-000000dc answered SIP/David-000000db
Audio is at 18872
Video is at xx.xx.xx.xx:10312
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding video codec h263 to SDP
Adding video codec h263p to SDP
Adding video codec vp8 to SDP
Adding video codec vp9 to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 103.77.138.124:6666 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-373635-4312c112d7c58e013374bf1765de1232;received=103.77.138.124;rport=6666
From: <sip:David@xx.xx.xx.xx>;tag=textClient
To: <sip:Smith@xx.xx.xx.xx>;tag=as4892a50a
Call-ID: 20e141e9fc949f37ba89730a2240c412@10.28.113.30
CSeq: 4 INVITE
Server: Asterisk PBX 16.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:852@xx.xx.xx.xx:5060>
Content-Type: application/sdp
Content-Length: 446

v=0
o=root 135677352 135677352 IN IP4 xx.xx.xx.xx
s=Asterisk PBX 16.7.0
c=IN IP4 xx.xx.xx.xx
b=CT:384
t=0 0
m=audio 18872 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 10312 RTP/AVP 34 103 100 108
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=rtpmap:100 VP8/90000
a=rtcp-fb:* ccm fir
a=rtpmap:108 VP9/90000
a=sendrecv

<------------>
    -- Channel SIP/Smith-000000dc joined 'simple_bridge' basic-bridge <e6f435ff-ee3c-48ce-bdf7-85c86dba88c3>
    -- Channel SIP/David-000000db joined 'simple_bridge' basic-bridge <e6f435ff-ee3c-48ce-bdf7-85c86dba88c3>

<--- SIP read from UDP:103.77.138.124:6666 --->
ACK sip:852@xx.xx.xx.xx:5060 SIP/2.0
Call-ID: 20e141e9fc949f37ba89730a2240c412@10.28.113.30
CSeq: 4 ACK
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-373635-e3677d66d035da8a8d3adeab7871177d
From: <sip:David@xx.xx.xx.xx>;tag=textClient
To: <sip:Smith@xx.xx.xx.xx>;tag=as4892a50a
Max-Forwards: 70
Authorization: Digest username="David",realm="asterisk",nonce="6f72a46b",uri="sip:852@xx.xx.xx.xx:5060;maddr=xx.xx.xx.xx",response="74baa7898d75d3ce409bb9f7e676e7a9",algorithm=MD5
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Got  RTP packet from    103.77.138.124:7078 (type 00, seq 000004, ts 3262933439, len 000160)
Got  RTP packet from    103.77.138.124:7078 (type 00, seq 000005, ts 3262933599, len 000160)
Got  RTP packet from    103.77.138.124:7078 (type 00, seq 000006, ts 3262933759, len 000160)
Got  RTP packet from    103.77.138.124:7078 (type 00, seq 000007, ts 3262933919, len 000160)

<--- SIP read from UDP:103.77.138.124:54622 --->


<------------->
Got  RTP packet from    103.77.138.124:7078 (type 00, seq 000008, ts 3262934079, len 000160)
Got  RTP packet from    103.77.138.124:7078 (type 00, seq 000009, ts 3262934239, len 000160)
Got  RTP packet from    103.77.138.124:7078 (type 00, seq 000010, ts 3262934399, len 000160)
Got  RTP packet from    103.77.138.124:7078 (type 00, seq 000011, ts 3262934559, len 000160)
Got  RTP packet from    103.77.138.124:7078 (type 00, seq 000012, ts 3262934719, len 000160)
Got  RTP packet from    103.77.138.124:7078 (type 00, seq 000013, ts 3262934879, len 000160)
Got  RTP packet from    103.77.138.124:7078 (type 00, seq 000014, ts 3262935039, len 000160)

<--- SIP read from UDP:103.77.138.124:45379 --->


<------------->
Got  RTP packet from    103.77.138.124:7078 (type 00, seq 000015, ts 3262935199, len 000160)
Got  RTP packet from    103.77.138.124:7078 (type 00, seq 000016, ts 3262935359, len 000160)
Got  RTP packet from    103.77.138.124:7078 (type 00, seq 000017, ts 3262935519, len 000160)
Got  RTP packet from    103.77.138.124:7078 (type 00, seq 000018, ts 3262935679, len 000160)
Got  RTP packet from    103.77.138.124:7078 (type 00, seq 000019, ts 3262935839, len 000160)
[Jul 24 08:16:19] NOTICE[25076]: manager.c:3512 authenticate: 195.154.31.140 tried to authenticate with nonexistent user 'cron'
[Jul 24 08:16:19] NOTICE[25076]: manager.c:3549 authenticate: 195.154.31.140 failed to authenticate as 'cron'
Got  RTP packet from    103.77.138.124:7078 (type 00, seq 000020, ts 3262935999, len 000160)
Got  RTP packet from    103.77.138.124:7078 (type 00, seq 000021, ts 3262936159, len 000160)
Got  RTP packet from    103.77.138.124:7078 (type 00, seq 000022, ts 3262936319, len 000160)
Got  RTP packet from    103.77.138.124:7078 (type 00, seq 000023, ts 3262936479, len 000160)
Got  RTP packet from    103.77.138.124:7078 (type 00, seq 000024, ts 3262936639, len 000160)
Got  RTP packet from    103.77.138.124:7078 (type 00, seq 000025, ts 3262936799, len 000160)
Got  RTP packet from    103.77.138.124:7078 (type 00, seq 000026, ts 3262936959, len 000160)

Any mistake I am doing? Please advice

Is every xx.xx.xx.xx the same? If not, there is no point in trying to analyze this for media problems, other than to ask whether you have opened the firewall for media.

We need to be able to distinguish between different addresses and to be able to distinguish each group of local addresses from public addresses.

Yes… The xx.xx.xx.xx are same.
Please find the details with Asterisk IP Address hosted at Cloud…

<--- SIP read from UDP:103.77.138.124:6666 --->
REGISTER sip:172.63.255.99:5060;maddr=172.63.255.99 SIP/2.0
Via: SIP/2.0/UDP 10.28.113.30:5060;branch=z9hG4bK-3633-ffabbbb3dcd1705a1567754ddb9af3ce
Max-Forwards: 70
From: <sip:David@172.63.255.99>;tag=textClient
To: <sip:David@172.63.255.99>
Call-ID: f73caf64144d2cc1d65a75f7648c3d5d@10.28.113.30
CSeq: 2 REGISTER
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Contact: <sip:David@10.28.113.30:6666;transport=udp>
Expires: 3060
Authorization: Digest username="David",realm="asterisk",nonce="7b4da517",uri="sip:172.63.255.99:5060;maddr=172.63.255.99",response="1541c6dbcb33e0f65e0e5fabd496b1ae",algorithm=MD5
Content-Length: 0
=============
--- (12 headers 0 lines) ---
Sending to 103.77.138.124:6666 (NAT)
Reliably Transmitting (NAT) to 103.77.138.124:6666:
OPTIONS sip:David@10.28.113.30:6666;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.63.255.99:5060;branch=z9hG4bK18781504;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.63.255.99>;tag=as78e5b4da
To: <sip:David@10.28.113.30:6666;transport=udp>
Contact: <sip:asterisk@172.63.255.99:5060>
Call-ID: 54a7b1f320190cbb52fa370b0c401023@172.63.255.99:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.7.0
Date: Fri, 24 Jul 2020 08:26:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
============<--- Transmitting (NAT) to 103.77.138.124:6666 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.28.113.30:5060;branch=z9hG4bK-3633-ffabbbb3dcd1705a1567754ddb9af3ce;received=103.77.138.124;rport=6666
From: <sip:David@172.63.255.99>;tag=textClient
To: <sip:David@172.63.255.99>;tag=as37e6c212
Call-ID: f73caf64144d2cc1d65a75f7648c3d5d@10.28.113.30
CSeq: 2 REGISTER
Server: Asterisk PBX 16.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 3060
Contact: <sip:David@10.28.113.30:6666;transport=udp>;expires=3060
Date: Fri, 24 Jul 2020 08:26:36 GMT
Content-Length: 0
===========<------------>
Scheduling destruction of SIP dialog 'f73caf64144d2cc1d65a75f7648c3d5d@10.28.113.30' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:103.77.138.124:6666 --->
SIP/2.0 200 OK
CSeq: 102 OPTIONS
Call-ID: 54a7b1f320190cbb52fa370b0c401023@172.63.255.99:5060
From: "asterisk" <sip:asterisk@172.63.255.99>;tag=as78e5b4da
To: <sip:David@10.28.113.30:6666;transport=udp>
Via: SIP/2.0/UDP 172.63.255.99:5060;branch=z9hG4bK18781504;rport=5060;received=172.63.255.99
Contact: <sip:David@10.28.113.30:6666;transport=udp>
Content-Length: 0
===============
--- (8 headers 0 lines) ---
Really destroying SIP dialog '54a7b1f320190cbb52fa370b0c401023@172.63.255.99:5060' Method: OPTIONS

<--- SIP read from UDP:103.77.138.124:6666 --->
INVITE sip:852@172.63.255.99 SIP/2.0
CSeq: 2 INVITE
To: <sip:Smith@172.63.255.99>
Call-ID: 9b37308a55a2fac98d8bc4864b7fd38f@10.28.113.30
From: <sip:David@172.63.255.99>;tag=textClient
Max-Forwards: 70
Contact: <sip:David@10.28.113.30:6666;transport=udp>
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-3633-d578df4ffa41c2ae90722d87b35f9a09
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 103.77.138.124:6666 (NAT)
Sending to 103.77.138.124:6666 (NAT)
Using INVITE request as basis request - 9b37308a55a2fac98d8bc4864b7fd38f@10.28.113.30
Found peer 'David' for 'David' from 103.77.138.124:6666

<--- Reliably Transmitting (NAT) to 103.77.138.124:6666 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-3633-d578df4ffa41c2ae90722d87b35f9a09;received=103.77.138.124;rport=6666
From: <sip:David@172.63.255.99>;tag=textClient
To: <sip:Smith@172.63.255.99>;tag=as53b9c771
Call-ID: 9b37308a55a2fac98d8bc4864b7fd38f@10.28.113.30
CSeq: 2 INVITE
Server: Asterisk PBX 16.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="606625aa"
Content-Length: 0


<------------>
<--- SIP read from UDP:103.77.138.124:6666 --->
ACK sip:852@172.63.255.99 SIP/2.0
Call-ID: 9b37308a55a2fac98d8bc4864b7fd38f@10.28.113.30
Max-Forwards: 70
From: <sip:David@172.63.255.99>;tag=textClient
To: <sip:Smith@172.63.255.99>;tag=as53b9c771
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-3633-d578df4ffa41c2ae90722d87b35f9a09
CSeq: 2 ACK
Content-Length: 0

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

<--- SIP read from UDP:103.77.138.124:6666 --->
INVITE sip:852@172.63.255.99:5060;maddr=172.63.255.99 SIP/2.0
CSeq: 3 INVITE
To: <sip:Smith@172.63.255.99>
Call-ID: 9b37308a55a2fac98d8bc4864b7fd38f@10.28.113.30
From: <sip:David@172.63.255.99>;tag=textClient
Max-Forwards: 70
Contact: <sip:David@10.28.113.30:6666;transport=udp>
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-3633-72c2c45d0aea33e399bf65e58455a805
Authorization: Digest username="David",realm="asterisk",nonce="606625aa",uri="sip:852@172.63.255.99:5060;maddr=172.63.255.99",response="104f273ac12d689c96c5d29e4a87aacc",algorithm=MD5
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 103.77.138.124:6666 (NAT)
Sending to 103.77.138.124:6666 (NAT)
Using INVITE request as basis request - 9b37308a55a2fac98d8bc4864b7fd38f@10.28.113.30
Found peer 'David' for 'David' from 103.77.138.124:6666

<--- Reliably Transmitting (NAT) to 103.77.138.124:6666 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-3633-72c2c45d0aea33e399bf65e58455a805;received=103.77.138.124;rport=6666
From: <sip:David@172.63.255.99>;tag=textClient
To: <sip:Smith@172.63.255.99>;tag=as7e2c5123
Call-ID: 9b37308a55a2fac98d8bc4864b7fd38f@10.28.113.30
CSeq: 3 INVITE
Server: Asterisk PBX 16.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="396b0012"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '9b37308a55a2fac98d8bc4864b7fd38f@10.28.113.30' in 11840 ms (Method: INVITE)

<--- SIP read from UDP:103.77.138.124:6666 --->
ACK sip:852@172.63.255.99:5060;maddr=172.63.255.99 SIP/2.0
Call-ID: 9b37308a55a2fac98d8bc4864b7fd38f@10.28.113.30
Max-Forwards: 70
From: <sip:David@172.63.255.99>;tag=textClient
To: <sip:Smith@172.63.255.99>;tag=as7e2c5123
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-3633-72c2c45d0aea33e399bf65e58455a805
CSeq: 3 ACK
Content-Length: 0

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

<--- SIP read from UDP:103.77.138.124:6666 --->
INVITE sip:852@172.63.255.99:5060;maddr=172.63.255.99 SIP/2.0
CSeq: 4 INVITE
To: <sip:Smith@172.63.255.99>
Call-ID: 9b37308a55a2fac98d8bc4864b7fd38f@10.28.113.30
From: <sip:David@172.63.255.99>;tag=textClient
Max-Forwards: 70
Contact: <sip:David@10.28.113.30:6666;transport=udp>
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-3633-27af22b3fa8adb8d033b120ea62b1c43
Authorization: Digest username="David",realm="asterisk",nonce="396b0012",uri="sip:852@172.63.255.99:5060;maddr=172.63.255.99",response="83233e7dd0ebae10fcf1f4facdde8cb2",algorithm=MD5
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 103.77.138.124:6666 (NAT)
Using INVITE request as basis request - 9b37308a55a2fac98d8bc4864b7fd38f@10.28.113.30
Found peer 'David' for 'David' from 103.77.138.124:6666
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
Looking for 852 in incoming (domain 172.63.255.99)
sip_route_dump: route/path hop: <sip:David@10.28.113.30:6666;transport=udp>

<--- Transmitting (NAT) to 103.77.138.124:6666 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-3633-27af22b3fa8adb8d033b120ea62b1c43;received=103.77.138.124;rport=6666
From: <sip:David@172.63.255.99>;tag=textClient
To: <sip:Smith@172.63.255.99>
Call-ID: 9b37308a55a2fac98d8bc4864b7fd38f@10.28.113.30
CSeq: 4 INVITE
Server: Asterisk PBX 16.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:852@172.63.255.99:5060>
Content-Length: 0


<------------>
    -- Executing [852@incoming:1] Dial("SIP/David-000000dd", "SIP/Smith")
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
Audio is at 13858
Video is at 172.63.255.99:10322
Adding codec ulaw to SDP
Adding video codec h263 to SDP
Adding video codec h263p to SDP
Adding video codec vp8 to SDP
Adding video codec vp9 to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 103.77.138.124:54622:
INVITE sip:Smith@103.77.138.124:54622;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.63.255.99:5060;branch=z9hG4bK757a12ba;rport
Max-Forwards: 70
From: <sip:David@172.63.255.99>;tag=as24fa164f
To: <sip:Smith@103.77.138.124:54622;transport=udp>
Contact: <sip:David@172.63.255.99:5060>
Call-ID: 582afea40f2f16120965162074049483@172.63.255.99:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.7.0
Date: Fri, 24 Jul 2020 08:26:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 446

v=0
o=root 826047459 826047459 IN IP4 172.63.255.99
s=Asterisk PBX 16.7.0
c=IN IP4 172.63.255.99
b=CT:384
t=0 0
m=audio 13858 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 10322 RTP/AVP 34 103 100 108
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=rtpmap:100 VP8/90000
a=rtcp-fb:* ccm fir
a=rtpmap:108 VP9/90000
a=sendrecv

---
---
  == Connect attempt from '195.154.31.140' unable to authenticate
Retransmitting #1 (NAT) to 103.77.138.124:54622:
INVITE sip:Smith@103.77.138.124:54622;transport=udp SIP/2.0
Via: SIP/2.0/UDP 172.63.255.99:5060;branch=z9hG4bK757a12ba;rport
Max-Forwards: 70
From: <sip:David@172.63.255.99>;tag=as24fa164f
To: <sip:Smith@103.77.138.124:54622;transport=udp>
Contact: <sip:David@172.63.255.99:5060>
Call-ID: 582afea40f2f16120965162074049483@172.63.255.99:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.7.0
Date: Fri, 24 Jul 2020 08:26:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 446

v=0
o=root 826047459 826047459 IN IP4 172.63.255.99
s=Asterisk PBX 16.7.0
c=IN IP4 172.63.255.99
b=CT:384
t=0 0
m=audio 13858 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 10322 RTP/AVP 34 103 100 108
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=rtpmap:100 VP8/90000
a=rtcp-fb:* ccm fir
a=rtpmap:108 VP9/90000
a=sendrecv

---

<--- SIP read from UDP:103.77.138.124:54622 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.63.255.99:5060;branch=z9hG4bK757a12ba;rport
From: <sip:David@172.63.255.99>;tag=as24fa164f
To: <sip:Smith@103.77.138.124:54622;transport=udp>
Call-ID: 582afea40f2f16120965162074049483@172.63.255.99:5060
CSeq: 102 INVITE

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

<--- SIP read from UDP:103.77.138.124:54622 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.63.255.99:5060;branch=z9hG4bK757a12ba;rport
From: <sip:David@172.63.255.99>;tag=as24fa164f
To: <sip:Smith@103.77.138.124:54622;transport=udp>;tag=P1vhbK~
Call-ID: 582afea40f2f16120965162074049483@172.63.255.99:5060
CSeq: 102 INVITE
User-Agent: LinphoneAndroid/4.3.0 (Galaxy M30s) LinphoneSDK/4.4.0 (tags/4.4.0^0)
Supported: replaces, outbound, gruu

<------------->
--- (8 headers 0 lines) ---
sip_route_dump: no route/path
    -- SIP/Smith-000000de is ringing

<--- Transmitting (NAT) to 103.77.138.124:6666 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-3633-27af22b3fa8adb8d033b120ea62b1c43;received=103.77.138.124;rport=6666
From: <sip:David@172.63.255.99>;tag=textClient
To: <sip:Smith@172.63.255.99>;tag=as5340f7c4
Call-ID: 9b37308a55a2fac98d8bc4864b7fd38f@10.28.113.30
CSeq: 4 INVITE
Server: Asterisk PBX 16.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:852@172.63.255.99:5060>
Content-Length: 0


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

<--- SIP read from UDP:103.77.138.124:54622 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.63.255.99:5060;branch=z9hG4bK757a12ba;rport
From: <sip:David@172.63.255.99>;tag=as24fa164f
To: <sip:Smith@103.77.138.124:54622;transport=udp>;tag=P1vhbK~
Call-ID: 582afea40f2f16120965162074049483@172.63.255.99:5060
CSeq: 102 INVITE
User-Agent: LinphoneAndroid/4.3.0 (Galaxy M30s) LinphoneSDK/4.4.0 (tags/4.4.0^0)
Supported: replaces, outbound, gruu

<------------->
--- (8 headers 0 lines) ---
sip_route_dump: no route/path
    -- SIP/Smith-000000de is ringing

The trace stops at a point where the call is still ringing and Asterisk has no idea where to send media to either side.

The A side has used late offer SDP, so won’t tell Asterisk where to send media until after the call has been answered. The B side hasn’t sent any media information in its provisional responses, so it seems it is not going to tell Asterisk where to send media until it signals the call is answered.

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