Voice Not received

Hello,
I am calling from Java Application [JAINSIP] to Linphone on Android for calling.
While the call is generated Android Linphone is ringing correctly. Now when I receive the call at Linphone, no voice is coming either side.

Below is my RTP log

== Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
    -- Executing [852@incoming:1] Dial("SIP/David-00000082", "SIP/Suman")
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
    -- Called SIP/Arijit
    -- SIP/Suman-00000083 is ringing
=============
Got  RTP packet from    103.77.138.75:7078 (type 00, seq 000004, ts 527566587, len 000160)
Got  RTP packet from    103.77.138.75:7078 (type 00, seq 000006, ts 527566907, len 000160)
Got  RTP packet from    103.77.138.75:7078 (type 00, seq 000007, ts 527567067, len 000160)
Got  RTP packet from    103.77.138.75:7078 (type 00, seq 000008, ts 527567227, len 000160)
Got  RTP packet from    103.77.138.75:7078 (type 00, seq 000009, ts 527567387, len 000160)
Got  RTP packet from    103.77.138.75:7078 (type 00, seq 000010, ts 527567547, len 000160)
Got  RTP packet from    103.77.138.75:7078 (type 00, seq 000011, ts 527567707, len 000160)
Got  RTP packet from    103.77.138.75:7078 (type 00, seq 000012, ts 527567867, len 000160)
Got  RTP packet from    103.77.138.75:7078 (type 00, seq 000013, ts 527568027, len 000160)
Got  RTP packet from    103.77.138.75:7078 (type 00, seq 000014, ts 527568187, len 000160)
Got  RTP packet from    103.77.138.75:7078 (type 00, seq 000015, ts 527568347, len 000160)
Got  RTP packet from    103.77.138.75:7078 (type 00, seq 000016, ts 527568507, len 000160)
Got  RTP packet from    103.77.138.75:7078 (type 00, seq 000017, ts 527568667, len 000160)
Got  RTP packet from    103.77.138.75:7078 (type 00, seq 000018, ts 527568827, len 000160)
Got  RTP packet from    103.77.138.75:7078 (type 00, seq 000019, ts 527568987, len 000160)
Got  RTP packet from    103.77.138.75:7078 (type 00, seq 000020, ts 527569147, len 000160)
Got  RTP packet from    103.77.138.75:7078 (type 00, seq 000021, ts 527569307, len 000160)
Got  RTP packet from    103.77.138.75:7078 (type 00, seq 000022, ts 527569467, len 000160)
Got  RTP packet from    103.77.138.75:7078 (type 00, seq 000023, ts 527569627, len 000160)
Got  RTP packet from    103.77.138.75:7078 (type 00, seq 000024, ts 527569787, len 000160)
Got  RTP packet from    103.77.138.75:7078 (type 00, seq 000025, ts 527569947, len 000160)
Got  RTP packet from    103.77.138.75:7078 (type 00, seq 000026, ts 527570107, len 000160)
Got  RTP packet from    103.77.138.75:7078 (type 00, seq 000027, ts 527570267, len 000160)

Am I doing any mistake? Please suggest.

The call hasn’t been answered, according to the log.

Please find the below log from SIP SET DEBUG ON

REGISTER sip:xx.xx.xx.xx SIP/2.0
Via: SIP/2.0/UDP 10.28.113.30:5060;branch=z9hG4bK-313933-b93333d08e0b9a53491e10cdc89d5ad4
Max-Forwards: 70
From: <sip:David@xx.xx.xx.xx>;tag=textClient
To: <sip:David@xx.xx.xx.xx>
Call-ID: 20510a35139af506ad9c1fe09fbd4838@10.28.113.30
CSeq: 1 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
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 103.77.138.102:6666 (NAT)
Sending to 103.77.138.102:6666 (NAT)

<--- Transmitting (NAT) to 103.77.138.102:6666 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.28.113.30:5060;branch=z9hG4bK-313933-b93333d08e0b9a53491e10cdc89d5ad4;received=103.77.138.102;rport=6666
From: <sip:David@xx.xx.xx.xx>;tag=textClient
To: <sip:David@xx.xx.xx.xx>;tag=as295b74c5
Call-ID: 20510a35139af506ad9c1fe09fbd4838@10.28.113.30
CSeq: 1 REGISTER
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="2f18cdfa"
Content-Length: 0
-------------
<--- SIP read from UDP:103.77.138.102:6666 --->
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-313933-d06ba05ec165f316f08c16fbc114ee5d
Max-Forwards: 70
From: <sip:David@xx.xx.xx.xx>;tag=textClient
To: <sip:David@xx.xx.xx.xx>
Call-ID: 20510a35139af506ad9c1fe09fbd4838@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="2f18cdfa",uri="sip:xx.xx.xx.xx:5060;maddr=xx.xx.xx.xx",response="3907394890bfdcfd321b93d9e14f2559",algorithm=MD5
Content-Length: 0

--- (12 headers 0 lines) ---
Sending to 103.77.138.102:6666 (NAT)
    -- Registered SIP 'David' at 103.77.138.102:6666
Reliably Transmitting (NAT) to 103.77.138.102:6666:
OPTIONS sip:David@10.28.113.30:6666;transport=udp SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK1dc2a308;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@xx.xx.xx.xx>;tag=as62f11c6a
To: <sip:David@10.28.113.30:6666;transport=udp>
Contact: <sip:asterisk@xx.xx.xx.xx:5060>
Call-ID: 7923d7b952fdfbf936a8b4586e5225a8@xx.xx.xx.xx:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.7.0
Date: Mon, 20 Jul 2020 11:46:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
-------------------
<--- Transmitting (NAT) to 103.77.138.102:6666 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.28.113.30:5060;branch=z9hG4bK-313933-d06ba05ec165f316f08c16fbc114ee5d;received=103.77.138.102;rport=6666
From: <sip:David@xx.xx.xx.xx>;tag=textClient
To: <sip:David@xx.xx.xx.xx>;tag=as295b74c5
Call-ID: 20510a35139af506ad9c1fe09fbd4838@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: Mon, 20 Jul 2020 11:46:26 GMT
Content-Length: 0
================
<--- SIP read from UDP:103.77.138.102:6666 --->
SIP/2.0 200 OK
CSeq: 102 OPTIONS
Call-ID: 7923d7b952fdfbf936a8b4586e5225a8@xx.xx.xx.xx:5060
From: "asterisk" <sip:asterisk@xx.xx.xx.xx>;tag=as62f11c6a
To: <sip:David@10.28.113.30:6666;transport=udp>
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK1dc2a308;rport=5060;received=xx.xx.xx.xx
Contact: <sip:David@10.28.113.30:6666;transport=udp>
Content-Length: 0

----------------------
<--- SIP read from UDP:103.77.138.102:6666 --->
INVITE sip:852@xx.xx.xx.xx SIP/2.0
CSeq: 2 INVITE
To: <sip:Suman@xx.xx.xx.xx>
Call-ID: 05e0a6d438701388a0fb13f457cab9e3@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-313933-badb3ebce253cff633cbd4370a4648fc
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 103.77.138.102:6666 (NAT)
Sending to 103.77.138.102:6666 (NAT)
Using INVITE request as basis request - 05e0a6d438701388a0fb13f457cab9e3@10.28.113.30
Found peer 'David' for 'David' from 103.77.138.102:6666

<--- Reliably Transmitting (NAT) to 103.77.138.102:6666 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-313933-badb3ebce253cff633cbd4370a4648fc;received=103.77.138.102;rport=6666
From: <sip:David@xx.xx.xx.xx>;tag=textClient
To: <sip:Suman@xx.xx.xx.xx>;tag=as5b22863f
Call-ID: 05e0a6d438701388a0fb13f457cab9e3@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="5038cc91"
Content-Length: 0
-------------
<--- SIP read from UDP:103.77.138.102:6666 --->
ACK sip:852@xx.xx.xx.xx SIP/2.0
Call-ID: 05e0a6d438701388a0fb13f457cab9e3@10.28.113.30
Max-Forwards: 70
From: <sip:David@xx.xx.xx.xx>;tag=textClient
To: <sip:Suman@xx.xx.xx.xx>;tag=as5b22863f
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-313933-badb3ebce253cff633cbd4370a4648fc
CSeq: 2 ACK
Content-Length: 0
--------------
<--- SIP read from UDP:103.77.138.102:6666 --->
INVITE sip:852@xx.xx.xx.xx:5060;maddr=xx.xx.xx.xx SIP/2.0
CSeq: 3 INVITE
To: <sip:Suman@xx.xx.xx.xx>
Call-ID: 05e0a6d438701388a0fb13f457cab9e3@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-313933-17de3781348ddce17fd108590df453dc
Authorization: Digest username="David",realm="asterisk",nonce="5038cc91",uri="sip:852@xx.xx.xx.xx:5060;maddr=xx.xx.xx.xx",response="3816b2d2542dd173da0cc113475dc0a4",algorithm=MD5
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 103.77.138.102:6666 (NAT)
Sending to 103.77.138.102:6666 (NAT)
Using INVITE request as basis request - 05e0a6d438701388a0fb13f457cab9e3@10.28.113.30
Found peer 'David' for 'David' from 103.77.138.102:6666

=============
<--- Reliably Transmitting (NAT) to 103.77.138.102:6666 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-313933-17de3781348ddce17fd108590df453dc;received=103.77.138.102;rport=6666
From: <sip:David@xx.xx.xx.xx>;tag=textClient
To: <sip:Suman@xx.xx.xx.xx>;tag=as35c81f02
Call-ID: 05e0a6d438701388a0fb13f457cab9e3@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="323db9e8"
Content-Length: 0
=================
<--- SIP read from UDP:103.77.138.102:6666 --->
ACK sip:852@xx.xx.xx.xx:5060;maddr=xx.xx.xx.xx SIP/2.0
Call-ID: 05e0a6d438701388a0fb13f457cab9e3@10.28.113.30
Max-Forwards: 70
From: <sip:David@xx.xx.xx.xx>;tag=textClient
To: <sip:Suman@xx.xx.xx.xx>;tag=as35c81f02
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-313933-17de3781348ddce17fd108590df453dc
CSeq: 3 ACK
Content-Length: 0

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

<--- SIP read from UDP:103.77.138.102:6666 --->
INVITE sip:852@xx.xx.xx.xx:5060;maddr=xx.xx.xx.xx SIP/2.0
CSeq: 4 INVITE
To: <sip:Suman@xx.xx.xx.xx>
Call-ID: 05e0a6d438701388a0fb13f457cab9e3@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-313933-59249074ecb8d16c36d1874f750069ac
Authorization: Digest username="David",realm="asterisk",nonce="323db9e8",uri="sip:852@xx.xx.xx.xx:5060;maddr=xx.xx.xx.xx",response="8f00344f491a3dfc870d276d15f0b2fa",algorithm=MD5
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 103.77.138.102:6666 (NAT)
Using INVITE request as basis request - 05e0a6d438701388a0fb13f457cab9e3@10.28.113.30
Found peer 'David' for 'David' from 103.77.138.102:6666
  == 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 103.77.138.102:6666 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-313933-59249074ecb8d16c36d1874f750069ac;received=103.77.138.102;rport=6666
From: <sip:David@xx.xx.xx.xx>;tag=textClient
To: <sip:Suman@xx.xx.xx.xx>
Call-ID: 05e0a6d438701388a0fb13f457cab9e3@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-00000027", "SIP/Suman")
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
Audio is at 14092
Video is at xx.xx.xx.xx:12902
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 103.77.138.102:54867:
INVITE sip:Suman@10.28.113.30:54867;ob SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK6619e683;rport
Max-Forwards: 70
From: <sip:David@xx.xx.xx.xx>;tag=as67d8ce42
To: <sip:Suman@10.28.113.30:54867;ob>
Contact: <sip:David@xx.xx.xx.xx:5060>
Call-ID: 34a9677679286f661c6f0396289a7054@xx.xx.xx.xx:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.7.0
Date: Mon, 20 Jul 2020 11:46:27 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 403845166 403845166 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 14092 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 12902 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

---
    -- Called SIP/Suman
Retransmitting #1 (NAT) to 103.77.138.102:54867:
INVITE sip:Suman@10.28.113.30:54867;ob SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK6619e683;rport
Max-Forwards: 70
From: <sip:David@xx.xx.xx.xx>;tag=as67d8ce42
To: <sip:Suman@10.28.113.30:54867;ob>
Contact: <sip:David@xx.xx.xx.xx:5060>
Call-ID: 34a9677679286f661c6f0396289a7054@xx.xx.xx.xx:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.7.0
Date: Mon, 20 Jul 2020 11:46:27 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 403845166 403845166 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 14092 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 12902 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:103.77.138.102:54867 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;rport=5060;received=xx.xx.xx.xx;branch=z9hG4bK6619e683
Call-ID: 34a9677679286f661c6f0396289a7054@xx.xx.xx.xx:5060
From: <sip:David@xx.xx.xx.xx>;tag=as67d8ce42
To: <sip:Suman@10.28.113.30;ob>
CSeq: 102 INVITE
Content-Length: 0

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

<--- SIP read from UDP:103.77.138.102:54867 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;rport=5060;received=xx.xx.xx.xx;branch=z9hG4bK6619e683
Call-ID: 34a9677679286f661c6f0396289a7054@xx.xx.xx.xx:5060
From: <sip:David@xx.xx.xx.xx>;tag=as67d8ce42
To: <sip:Suman@10.28.113.30;ob>;tag=940f53ecb10f47ab86d86af2ed193e10
CSeq: 102 INVITE
Contact: <sip:Suman@10.28.113.30:54867;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:Suman@10.28.113.30:54867;ob>
    -- SIP/Suman-00000028 is ringing

<--- Transmitting (NAT) to 103.77.138.102:6666 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-313933-59249074ecb8d16c36d1874f750069ac;received=103.77.138.102;rport=6666
From: <sip:David@xx.xx.xx.xx>;tag=textClient
To: <sip:Suman@xx.xx.xx.xx>;tag=as366989b2
Call-ID: 05e0a6d438701388a0fb13f457cab9e3@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


<------------>
  == Connect attempt from '51.11.185.23' unable to authenticate

<--- SIP read from UDP:103.77.138.102:54867 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;rport=5060;received=xx.xx.xx.xx;branch=z9hG4bK6619e683
Call-ID: 34a9677679286f661c6f0396289a7054@xx.xx.xx.xx:5060
From: <sip:David@xx.xx.xx.xx>;tag=as67d8ce42
To: <sip:Suman@10.28.113.30;ob>;tag=940f53ecb10f47ab86d86af2ed193e10
CSeq: 102 INVITE
Contact: <sip:Suman@10.28.113.30:54867;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0

<------------->
Reliably Transmitting (NAT) to 103.77.138.102:54867:
OPTIONS sip:Suman@10.28.113.30:54867;ob SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK23ec596c;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@xx.xx.xx.xx>;tag=as16f969c7
To: <sip:Suman@10.28.113.30:54867;ob>
Contact: <sip:asterisk@xx.xx.xx.xx:5060>
Call-ID: 6e742ab61b825f59074509335ceff2f5@xx.xx.xx.xx:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.7.0
Date: Mon, 20 Jul 2020 11:46:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
<--- SIP read from UDP:103.77.138.102:54867 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;rport=5060;received=xx.xx.xx.xx;branch=z9hG4bK23ec596c
Call-ID: 6e742ab61b825f59074509335ceff2f5@xx.xx.xx.xx:5060
From: "asterisk" <sip:asterisk@xx.xx.xx.xx>;tag=as16f969c7
To: <sip:Suman@10.28.113.30;ob>;tag=z9hG4bK23ec596c
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: MicroSIP/3.19.10
Content-Length: 0
===============
<--- SIP read from UDP:103.77.138.102:54867 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;rport=5060;received=xx.xx.xx.xx;branch=z9hG4bK6619e683
Call-ID: 34a9677679286f661c6f0396289a7054@xx.xx.xx.xx:5060
From: <sip:David@xx.xx.xx.xx>;tag=as67d8ce42
To: <sip:Suman@10.28.113.30;ob>;tag=940f53ecb10f47ab86d86af2ed193e10
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: <sip:Suman@10.28.113.30:54867;ob>
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 420

v=0
o=- 3804254185 3804254187 IN IP4 10.28.113.30
s=pjmedia
b=AS:352
t=0 0
a=X-nat:0
m=audio 10008 RTP/AVP 0 101
c=IN IP4 10.28.113.30
b=TIAS:64000
a=rtcp:10009 IN IP4 10.28.113.30
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1093611761 cname:6ea14c665c5e6d4e
m=video 0 RTP/AVP 103
c=IN IP4 10.28.113.30
a=rtpmap:103 H263-1998/90000
a=fmtp:103 CIF=1;QCIF=1
<------------->
--- (11 headers 19 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|h263|h263p|vp8|vp9), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.28.113.30:10008
Peer doesn't provide video
sip_route_dump: route/path hop: <sip:Suman@10.28.113.30:54867;ob>
Transmitting (NAT) to 103.77.138.102:54867:
ACK sip:Suman@10.28.113.30:54867;ob SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK7c5f5fd7;rport
Max-Forwards: 70
From: <sip:David@xx.xx.xx.xx>;tag=as67d8ce42
To: <sip:Suman@10.28.113.30:54867;ob>;tag=940f53ecb10f47ab86d86af2ed193e10
Contact: <sip:David@xx.xx.xx.xx:5060>
Call-ID: 34a9677679286f661c6f0396289a7054@xx.xx.xx.xx:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.7.0
Content-Length: 0

---
    -- SIP/Suman-00000028 answered SIP/David-00000027
Audio is at 16126
Video is at xx.xx.xx.xx:11272
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 103.77.138.102:6666 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-313933-59249074ecb8d16c36d1874f750069ac;received=103.77.138.102;rport=6666
From: <sip:David@xx.xx.xx.xx>;tag=textClient
To: <sip:Suman@xx.xx.xx.xx>;tag=as366989b2
Call-ID: 05e0a6d438701388a0fb13f457cab9e3@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 381568014 381568014 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 16126 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 11272 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/Suman-00000028 joined 'simple_bridge' basic-bridge <aca0aaa1-e796-4dd0-8b0e-2c4b8dfa886b>
    -- Channel SIP/David-00000027 joined 'simple_bridge' basic-bridge <aca0aaa1-e796-4dd0-8b0e-2c4b8dfa886b>
Retransmitting #1 (NAT) to 103.77.138.102:6666:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-313933-59249074ecb8d16c36d1874f750069ac;received=103.77.138.102;rport=6666
From: <sip:David@xx.xx.xx.xx>;tag=textClient
To: <sip:Suman@xx.xx.xx.xx>;tag=as366989b2
Call-ID: 05e0a6d438701388a0fb13f457cab9e3@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 381568014 381568014 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 16126 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 11272 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

---
Retransmitting #2 (NAT) to 103.77.138.102:6666:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-313933-59249074ecb8d16c36d1874f750069ac;received=103.77.138.102;rport=6666
From: <sip:David@xx.xx.xx.xx>;tag=textClient
To: <sip:Suman@xx.xx.xx.xx>;tag=as366989b2
Call-ID: 05e0a6d438701388a0fb13f457cab9e3@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 381568014 381568014 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 16126 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 11272 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
----------------------------------------
Retransmitting #3 (NAT) to 103.77.138.102:6666:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.28.113.30:6666;branch=z9hG4bK-313933-59249074ecb8d16c36d1874f750069ac;received=103.77.138.102;rport=6666
From: <sip:David@xx.xx.xx.xx>;tag=textClient
To: <sip:Suman@xx.xx.xx.xx>;tag=as366989b2
Call-ID: 05e0a6d438701388a0fb13f457cab9e3@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 381568014 381568014 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 16126 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 11272 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

------------------------

Please note that I am calling from my desktop Java Application from [David] to [Suman]
Any Mistake I am doing? Please advice.

The Contact header is wrong in this request. One would expect it to be a public address, like the actual source address. This needs fixing at the the other end. A similar problem going the other way is why you end up with retransmissions.

Also, given that the from tag appears to be hard coded, it looks like you are trying to write your own SIP client and taking short cuts.

Also this log does include an indication that the call was answered, but is missing the RTP debugging, so we can’t see if audio is arriving.

When you are redacting addresses, please keep the distinction between different addresses and between public and private ones, as I’m seeing a lot of xxx.xxx.xxx.xxx, but suspect they may not all be the same.

Do you suspect that this is a issue of NAT from SIP Client? Also I have uploaded the RTP Debug log at my first
post.

Please advice

The peer doesn’t seem to be aware that
it is behind NAT, both when acting as client (INVITE) and as server (response).