Asterisk 15 chan_sip TLS SRTP - no audio no RTP packets between IP Phones

Hello everyone,
I have read and tested a lot of things before opening this topic, based on previous problems and solutions on the “no audio issue” in Asterisk, but my problem still persists, so wanted to see if any additional idea from the Asterisk Community why it’s happening in my case.
I am implementing Secure Calls ( SRTP media traffic and TLS Signaling) on an Asterisk 15.4 VoIP Server. When using RTP/UDP without Encryption the audio calls are OK - no issues, after moving to Encrypted calls then no audio and I see no packets received on the 2 Cisco SPA504G IP Phones after call established, even they have transmitted a lot of packets during the calls as we can see from pics below.
I have monitored with SIP set debug on and RTP debug on and it looks there is transmission/receive of packets on the Asterisk CLI, but still no voice/audio.
I have opened the RTP port range 8000-65000 on rtp.conf and configured as per asterisk community suggestions about encryption=yes, transport=tls for SIP peers/EXTs and certificates which are installed on the IP Phones successfully.

There is no NAT implemented as they are internal LANs/WAN Calls inside an internal Network.

sip.conf

[general]
bindaddr=10.30.150.3
allowguest=no
callcounter=yes
srvlookup=no
trustrpid=no
sendrpid=no
disallow=all
allow=alaw,ulaw,gsm,g729
qualify=no
buggymwi=yes
rtptimeout=60
rtpholdtimeout=300
nat=force_rport,comedia

;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
encryption=yes
tlsenable=yes
tlsbindaddr=0.0.0.0
transport=udp,tls,tcp
tcpenable=yes
tlscertfile=/tftpboot/Certifikatat/asterisk.pem
tlscafile=/tftpboot/Certifikatat/ca.crt
tlsdontverifyserver=yes
tlscipher=ALL
tlsclientmethod=tlsv1
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

Asterisk 15 is end of life.

Please provide logs as plain text, not images.

Please provide protocol traces for both legs.

Where does 16509 come from?

Why do you have nat=force_rport,comedia if there is not NAT to cause a problem to work round?

chan_sip is no longer supported.

Hi David, thanks for the response and help

here is the trace from a SIP call between a Cisco IP Phone and a SoftPhone:

<— SIP read from TLS:10.30.150.139:5062 —>
SIP/2.0 200 OK
To: sip:00a2895ebf3b@10.30.150.139:5062;tag=3b7ad0f05c3bf9c1i0
From: “EkoF Encrypted Line1” sip:4930@10.30.150.3;tag=as5326b02c
Call-ID: 6770eea42d35774c73f335c6059f57de@10.30.150.3:5061
CSeq: 102 INVITE
Via: SIP/2.0/TLS 10.30.150.3:5061;branch=z9hG4bK66ec2c47
Contact: “EkoF 2 Encrypted Line2” sip:00a2895ebf3b@10.30.150.139:5062;transport=tls
Server: Cisco/SPA303-7.5.2
Content-Length: 297
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp

v=0
o=- 26534770 26534770 IN IP4 10.30.150.139
s=-
c=IN IP4 10.30.150.139
t=0 0
m=audio 16520 RTP/SAVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:hS+BHZu4v3grj5zZ8+L27CjTq/XsGtKAIDllcRvm
<------------->
— (12 headers 12 lines) —
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|ulaw|gsm|g729|h263p|h263|h264|h261|vp8|vp9), 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|)
> 0x7fb0d8008630 – Strict RTP learning after remote address set to: 10.30.150.139:16520
Peer audio RTP is at port 10.30.150.139:16520
sip_route_dump: route/path hop: sip:00a2895ebf3b@10.30.150.139:5062;transport=tls
Transmitting (NAT) to 10.30.150.139:5062:
ACK sip:00a2895ebf3b@10.30.150.139:5062;transport=tls SIP/2.0
Via: SIP/2.0/TLS 10.30.150.3:5061;branch=z9hG4bK3b67f4ca;rport
Max-Forwards: 70
From: “EkoF Encrypted Line1” sip:4930@10.30.150.3;tag=as5326b02c
To: sip:00a2895ebf3b@10.30.150.139:5062;transport=tls;tag=3b7ad0f05c3bf9c1i0
Contact: sip:4930@10.30.150.3:5061;transport=tls
Call-ID: 6770eea42d35774c73f335c6059f57de@10.30.150.3:5061
CSeq: 102 ACK
User-Agent: Asterisk PBX 15.4.1
Content-Length: 0


-- SIP/00a2895ebf3b-0000006e answered SIP/a44c119eb1f6-0000006d

Audio is at 37278
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding codec gsm to SDP
Adding codec g729 to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 192.168.5.46:49288 —>
SIP/2.0 200 OK
Via: SIP/2.0/TLS 192.168.5.46:49288;branch=z9hG4bK00aec2672676ed118e0eae45bd8f0b63;received=192.168.5.46;rport=49288
From: sip:a44c119eb1f6@10.30.150.3;tag=2383956475
To: sip:4931@10.30.150.3;tag=as44d6bc51
Call-ID: 00AEC267-2676-ED11-8E0C-AE45BD8F0B63@192.168.5.46
CSeq: 2 INVITE
Server: Asterisk PBX 15.4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:4931@10.30.150.3:5061;transport=tls
Content-Type: application/sdp
Content-Length: 414

v=0
o=root 295301950 295301950 IN IP4 10.30.150.3
s=Asterisk PBX 15.4.1
c=IN IP4 10.30.150.3
t=0 0
m=audio 37278 RTP/SAVP 8 0 3 18 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:hVY6Vf5luktureplGhh1R7ffPIwf9CvxFhxi+qsn
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

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

<— SIP read from TLS:192.168.5.46:49288 —>
ACK sip:4931@10.30.150.3:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 192.168.5.46:49288;branch=z9hG4bK00dbf3682676ed118e0eae45bd8f0b63;rport
From: sip:a44c119eb1f6@10.30.150.3;tag=2383956475
To: sip:4931@10.30.150.3;tag=as44d6bc51
Call-ID: 00AEC267-2676-ED11-8E0C-AE45BD8F0B63@192.168.5.46
CSeq: 2 ACK
Contact: sip:a44c119eb1f6@192.168.5.46:49288;transport=tls;gr=8061A71C-C073-ED11-8853-AE45BD8F0B63
Authorization: Digest username=“a44c119eb1f6”, realm=“asterisk”, nonce=“1b405930”, uri=“sip:4931@10.30.150.3:5061;transport=tls”, response=“84982557c1689386bad18ac2d2617109”, algorithm=MD5
Max-Forwards: 70
Content-Length: 0

<------------->
— (10 headers 0 lines) —
– Channel SIP/00a2895ebf3b-0000006e joined ‘simple_bridge’ basic-bridge
– Channel SIP/a44c119eb1f6-0000006d joined ‘simple_bridge’ basic-bridge
> 0x7fb0e401d290 – Strict RTP switching to RTP target address 192.168.5.46:5063 as source
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000001, ts 005390, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010907, ts 005384, len 000170)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000002, ts 005550, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010908, ts 005544, len 000170)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000003, ts 005710, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010909, ts 005704, len 000170)
> 0x7fb0d8008630 – Strict RTP switching to RTP target address 10.30.150.139:16520 as source
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004616, ts 77655001, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028447, ts 77655000, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000004, ts 005870, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010910, ts 005864, len 000170)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000005, ts 006030, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010911, ts 006024, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004617, ts 77655241, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028448, ts 77655240, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000006, ts 006190, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010912, ts 006184, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004618, ts 77655481, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028449, ts 77655480, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000007, ts 006350, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010913, ts 006344, len 000170)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000008, ts 006510, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010914, ts 006504, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004619, ts 77655721, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028450, ts 77655720, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000009, ts 006670, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010915, ts 006664, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004620, ts 77655961, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028451, ts 77655960, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000010, ts 006830, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010916, ts 006824, len 000170)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000011, ts 006990, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010917, ts 006984, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004621, ts 77656201, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028452, ts 77656200, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000012, ts 007150, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010918, ts 007144, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004622, ts 77656441, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028453, ts 77656440, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000013, ts 007310, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010919, ts 007304, len 000170)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000014, ts 007470, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010920, ts 007464, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004623, ts 77656681, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028454, ts 77656680, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000015, ts 007630, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010921, ts 007624, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004624, ts 77656921, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028455, ts 77656920, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000016, ts 007790, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010922, ts 007784, len 000170)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000017, ts 007950, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010923, ts 007944, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004625, ts 77657161, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028456, ts 77657160, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000018, ts 008110, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010924, ts 008104, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004626, ts 77657401, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028457, ts 77657400, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000019, ts 008270, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010925, ts 008264, len 000170)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000020, ts 008430, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010926, ts 008424, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004627, ts 77657641, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028458, ts 77657640, len 000250)

Got RTP packet from 192.168.5.46:5063 (type 08, seq 000021, ts 008590, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010927, ts 008584, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004628, ts 77657881, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028459, ts 77657880, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000022, ts 008750, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010928, ts 008744, len 000170)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000023, ts 008910, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010929, ts 008904, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004629, ts 77658121, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028460, ts 77658120, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000024, ts 009070, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010930, ts 009064, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004630, ts 77658361, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028461, ts 77658360, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000025, ts 009230, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010931, ts 009224, len 000170)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000026, ts 009390, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010932, ts 009384, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004631, ts 77658601, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028462, ts 77658600, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000027, ts 009550, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010933, ts 009544, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004632, ts 77658841, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028463, ts 77658840, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000028, ts 009710, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010934, ts 009704, len 000170)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000029, ts 009870, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010935, ts 009864, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004633, ts 77659081, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028464, ts 77659080, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000030, ts 010030, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010936, ts 010024, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004634, ts 77659321, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028465, ts 77659320, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000031, ts 010190, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010937, ts 010184, len 000170)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000032, ts 010350, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010938, ts 010344, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004635, ts 77659561, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028466, ts 77659560, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000033, ts 010510, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010939, ts 010504, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004636, ts 77659801, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028467, ts 77659800, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000034, ts 010670, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010940, ts 010664, len 000170)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000035, ts 010830, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010941, ts 010824, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004637, ts 77660041, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028468, ts 77660040, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000036, ts 010990, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010942, ts 010984, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004638, ts 77660281, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028469, ts 77660280, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000037, ts 011150, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010943, ts 011144, len 000170)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000038, ts 011310, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010944, ts 011304, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004639, ts 77660521, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028470, ts 77660520, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000039, ts 011470, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010945, ts 011464, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004640, ts 77660761, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028471, ts 77660760, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000040, ts 011630, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010946, ts 011624, len 000170)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000041, ts 011790, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010947, ts 011784, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004641, ts 77661001, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028472, ts 77661000, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000042, ts 011950, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010948, ts 011944, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004642, ts 77661241, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028473, ts 77661240, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000043, ts 012110, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010949, ts 012104, len 000170)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000044, ts 012270, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010950, ts 012264, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004643, ts 77661481, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028474, ts 77661480, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000045, ts 012430, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010951, ts 012424, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004644, ts 77661721, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028475, ts 77661720, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000046, ts 012590, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010952, ts 012584, len 000170)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000047, ts 012750, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010953, ts 012744, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004645, ts 77661961, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028476, ts 77661960, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000048, ts 012910, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010954, ts 012904, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004646, ts 77662201, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028477, ts 77662200, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000049, ts 013070, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010955, ts 013064, len 000170)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000050, ts 013230, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010956, ts 013224, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004647, ts 77662441, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028478, ts 77662440, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000051, ts 013390, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010957, ts 013384, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004648, ts 77662681, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028479, ts 77662680, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000052, ts 013550, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010958, ts 013544, len 000170)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000053, ts 013710, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010959, ts 013704, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004649, ts 77662921, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028480, ts 77662920, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000054, ts 013870, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010960, ts 013864, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004650, ts 77663161, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028481, ts 77663160, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000055, ts 014030, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010961, ts 014024, len 000170)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000056, ts 014190, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010962, ts 014184, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004651, ts 77663401, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028482, ts 77663400, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000057, ts 014350, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010963, ts 014344, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004652, ts 77663641, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028483, ts 77663640, len 000250)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000058, ts 014510, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010964, ts 014504, len 000170)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000059, ts 014670, len 000160)
Sent RTP packet to 10.30.150.139:16520 (type 08, seq 010965, ts 014664, len 000170)
Got RTP packet from 10.30.150.139:16520 (type 08, seq 004653, ts 77663881, len 000240)
Sent RTP packet to 192.168.5.46:5063 (type 08, seq 028484, ts 77663880, len 000250)

<— SIP read from TLS:10.30.150.139:5062 —>
BYE sip:4930@10.30.150.3:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 10.30.150.139:5062;branch=z9hG4bK-344a7960
From: sip:00a2895ebf3b@10.30.150.139;transport=tls;tag=3b7ad0f05c3bf9c1i0
To: “EkoF Encrypted Line1” sip:4930@10.30.150.3;tag=as5326b02c
Call-ID: 6770eea42d35774c73f335c6059f57de@10.30.150.3:5061
CSeq: 101 BYE
Max-Forwards: 70
User-Agent: Cisco/SPA303-7.5.2
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to 10.30.150.139:5062 (NAT)
Got RTP packet from 192.168.5.46:5063 (type 08, seq 000060, ts 014830, len 000160)
Scheduling destruction of SIP dialog ‘6770eea42d35774c73f335c6059f57de@10.30.150.3:5061’ in 6400 ms (Method: BYE)

<— Transmitting (NAT) to 10.30.150.139:5062 —>
SIP/2.0 200 OK
Via: SIP/2.0/TLS 10.30.150.139:5062;branch=z9hG4bK-344a7960;received=10.30.150.139;rport=5062
From: sip:00a2895ebf3b@10.30.150.139;transport=tls;tag=3b7ad0f05c3bf9c1i0
To: “EkoF Encrypted Line1” sip:4930@10.30.150.3;tag=as5326b02c
Call-ID: 6770eea42d35774c73f335c6059f57de@10.30.150.3:5061
CSeq: 101 BYE
Server: Asterisk PBX 15.4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
– Channel SIP/00a2895ebf3b-0000006e left ‘simple_bridge’ basic-bridge
– Channel SIP/a44c119eb1f6-0000006d left ‘simple_bridge’ basic-bridge
== Spawn extension (macro-stduser, s, 2) exited non-zero on ‘SIP/a44c119eb1f6-0000006d’ in macro ‘stduser’
== Spawn extension (KosoveMobil, 4931, 1) exited non-zero on ‘SIP/a44c119eb1f6-0000006d’
Scheduling destruction of SIP dialog ‘00AEC267-2676-ED11-8E0C-AE45BD8F0B63@192.168.5.46’ in 6400 ms (Method: ACK)
Reliably Transmitting (NAT) to 192.168.5.46:49288:
BYE sip:a44c119eb1f6@192.168.5.46:49288;transport=tls;gr=8061A71C-C073-ED11-8853-AE45BD8F0B63 SIP/2.0
Via: SIP/2.0/TLS 10.30.150.3:5061;branch=z9hG4bK25f56b02;rport
Max-Forwards: 70
From: sip:4931@10.30.150.3;tag=as44d6bc51
To: sip:a44c119eb1f6@10.30.150.3;tag=2383956475
Call-ID: 00AEC267-2676-ED11-8E0C-AE45BD8F0B63@192.168.5.46
CSeq: 102 BYE
User-Agent: Asterisk PBX 15.4.1
Proxy-Authorization: Digest username=“4930”, realm=“asterisk”, algorithm=MD5, uri=“sips:10.30.150.3”, nonce=“1b405930”, response=“41d874bac82d2fd04e61e2fb3bf0c67a”
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<— SIP read from TLS:192.168.5.46:49288 —>
SIP/2.0 200 OK
Via: SIP/2.0/TLS 10.30.150.3:5061;branch=z9hG4bK25f56b02;rport=5061
From: sip:4931@10.30.150.3;tag=as44d6bc51
To: sip:a44c119eb1f6@10.30.150.3;tag=2383956475
Call-ID: 00AEC267-2676-ED11-8E0C-AE45BD8F0B63@192.168.5.46
CSeq: 102 BYE
Contact: sip:a44c119eb1f6@192.168.5.46:5062;transport=tls;gr=8061A71C-C073-ED11-8853-AE45BD8F0B63
Server: phoner
Content-Length: 0

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

Where does 16509 come from?-

—About port 16509 i think it’s a port used by cisco IP Phones for rtp because i see on their config they use port range 16384 - 16538

Why do you have nat=force_rport,comedia if there is not NAT to cause a problem to work round?

  • I put nat=no but asterisk was suggesting to use force_rport,comedia instead.

chan_sip is no longer supported.

  • I know it and in the future (2-3 years) i plan to build a new server with latest Asterisk version (using Rocky Linux) and implement PJSIP but for the moment it’s impossible so I am trying to make it working using this version.

The point was where in the protocol exchange was anyone told to send to that port.

The correct thing to do, in most cases, is not to set it at all, which is equivalent to both settings being auto.

I don’t think somewhere it was told to send to that port, but when did the tcpdump on interface it was showing every few seconds some udp pot unreachable warnings like the one with 16509.

About audio issue I have tried without NAT part configuration, still no audio on the VoIP calls. Any additional idea why it’s happening ?

1 Like

Hello, did you resolve the problem ? I’m in the same situation, when I enable SRTP there is no audio. I have an online VPS, not a local server like you but still the same problem.

Hi, No I didn’t resolved the issue yet, doing some additional tests/configs and changes trying to solve it, also searching for different ideas and answers on internet but until now no solution for this scenario.

1 Like

Good news !!! :smiley: I found it.

I put nat=comedia and it worked. In fact, I was trying different combination of settings “nat” and “directmedia”, until I put nat="comedia", after that, I replaced it by "nat=auto_comedia" and it is still working, audio and video.

To resume my problem, the audio and video calls were working well without SRTP encryption, when I enabled encryption=yes and set SRTP as mendatory in the SIP software, the signalling worked, but when I answered the call, impossible to hear user speaking.

here is my settings in case:

using chan_sip with asterisk v13
TLS from let’s encrypt
VPS ubuntu 18 not behind a nat

Tests made with MicroSIP software on two laptops, one connected on my home internet, and the other on the hotspot of my smartphone, I used my cellphone data network to be sure that the problem was not related that both laptops were connected on same home router, and to simulate real situation, cause my server will be used by users who can be eveywhere.

sip.conf


[general]
context=visiochat
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=yes
tcpbindaddr=0.0.0.0
transport=tls
srvlookup=yes
videosupport=yes
allowguest=no
nat=auto_comedia
directmedia=update,nonat
tlsenable=yes
tlsbindaddr=0.0.0.0
tlsbindport=5061
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/fullchain.pem
tlscipher=ALL
tlsclientmethod=ALL
fromdomain=sip.mywebsiteaddress.com

users.conf

[default_template](!)              
type=friend                     
host=dynamic                       
dtmfmode=rfc2833                 
disallow=all                       
allow=ulaw                  
allow=h263p
allow=h264
allow=vp8
videosupport=yes
hassip=yes                 
hasiax=no                       
callwaiting = yes        
transfer=no          
canpark=no          
hasvoicemail=no       
qualify=yes        
encryption=yes
**************** VisioChat *****************
[1989](default_template)        
fullname = Ed Bowman     
username = Ed            
secret=1989
context=visiochat           
callerid="Ed Bowman"<1989>
transport=tls,udp,tcp

[3392](default_template)     
fullname = Alain Aldric     
username = aldric          
secret=aldric               
context=visiochat     
transport=tls,udp,tcp

Great to see that you are on the good way to solve the issue, please keep us updated if everything OK at the end

For now, everything’s okay, tested with my laptops and my android 11 phone (it supports SIP natively). That was my first step. The next step for me is WebRTC.

So now there is no problem with “no audio issue” using SRTP and TLS right?

Exactly.

1 Like

ok good, so please tell me if possible which is your final configuration on sip.conf, can you copy here your working version with SRTP enabled and TLS, which results with no audio issues?

It is the one I sent earlier, here it is:

[general]
context=visiochat
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=yes
tcpbindaddr=0.0.0.0
transport=tls
srvlookup=yes
videosupport=yes
allowguest=no
nat=auto_comedia
directmedia=update,nonat
tlsenable=yes
tlsbindaddr=0.0.0.0
tlsbindport=5061
tlscertfile=/etc/asterisk/keys/asterisk.pem
tlscafile=/etc/asterisk/keys/fullchain.pem
tlscipher=ALL
tlsclientmethod=ALL
fromdomain=sip.mywebsiteaddress.com

users.conf

[default_template](!)              
type=friend                     
host=dynamic                       
dtmfmode=rfc2833                 
disallow=all                       
allow=ulaw                  
allow=h263p
allow=h264
allow=vp8
videosupport=yes
hassip=yes                 
hasiax=no                       
callwaiting = yes        
transfer=no          
canpark=no          
hasvoicemail=no       
qualify=yes        
encryption=yes
**************** VisioChat *****************
[1989](default_template)        
fullname = Ed Bowman     
username = Ed            
secret=1989
context=visiochat           
callerid="Ed Bowman"<1989>
transport=tls

[3392](default_template)     
fullname = Alain Aldric     
username = aldric          
secret=aldric               
context=visiochat     
transport=tls

Everything in my sip.conf seems to be Fine - as per professional community forums suggestions like this one here in Asterisk community, and when I try the call everything looks OK on the SIP call flow, but still audio is missing on the call…strange situation.

The complete dialog of a Call:

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

— (12 headers 0 lines) —

Really destroying SIP dialog ‘57bcc0c13cc104e91e5514721ce3b334@10.30.150.3:5060’ Method: OPTIONS

<— SIP read from TLS:192.168.5.46:49804 —>

INVITE sip:4931@10.30.150.3 SIP/2.0

Via: SIP/2.0/TLS 192.168.5.46:49804;branch=z9hG4bK800b0eeaf28fed118173d91acb17e8b1;rport

From: sip:a44c119eb1f6@10.30.150.3;tag=1241913889

To: sip:4931@10.30.150.3

Call-ID: 800B0EEA-F28F-ED11-8172-D91ACB17E8B1@192.168.5.46

CSeq: 1 INVITE

Contact: sip:a44c119eb1f6@192.168.5.46:49804;transport=tls;gr=802F69CB-748F-ED11-806C-D91ACB17E8B1

Content-Type: application/sdp

Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK

Max-Forwards: 70

Supported: 100rel, replaces, from-change, gruu

User-Agent: phoner 3.23

P-Preferred-Identity: sip:a44c119eb1f6@10.30.150.3

Content-Length: 676

v=0

o=- 663715420 1 IN IP4 192.168.5.46

s=phoner 3.23

c=IN IP4 192.168.5.46

t=0 0

m=audio 5063 RTP/AVP 8 0 2 3 97 9 18 111 112 113 114 11 118 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:3 GSM/8000

a=rtpmap:97 iLBC/8000

a=rtpmap:9 G722/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=yes

a=rtpmap:111 speex/16000

a=rtpmap:112 G726-16/8000

a=rtpmap:113 G726-24/8000

a=rtpmap:114 G726-40/8000

a=rtpmap:11 L16/44100

a=rtpmap:118 L16/16000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:M6ZTEkROmlEgl4EI92wCMlMhps/MiPI/w510EiUf

a=encryption:optional

a=ssrc:1103747463

a=sendrecv

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

— (14 headers 26 lines) —

Sending to 192.168.5.46:49804 (no NAT)

Sending to 192.168.5.46:49804 (no NAT)

Using INVITE request as basis request - 800B0EEA-F28F-ED11-8172-D91ACB17E8B1@192.168.5.46

Found peer ‘a44c119eb1f6’ for ‘a44c119eb1f6’ from 192.168.5.46:49804

<— Reliably Transmitting (NAT) to 192.168.5.46:49804 —>

SIP/2.0 401 Unauthorized

Via: SIP/2.0/TLS 192.168.5.46:49804;branch=z9hG4bK800b0eeaf28fed118173d91acb17e8b1;received=192.168.5.46;rport=49804

From: sip:a44c119eb1f6@10.30.150.3;tag=1241913889

To: sip:4931@10.30.150.3;tag=as5a8db6cc

Call-ID: 800B0EEA-F28F-ED11-8172-D91ACB17E8B1@192.168.5.46

CSeq: 1 INVITE

Server: Asterisk PBX 15.4.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“40c42f29”

Content-Length: 0

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

Scheduling destruction of SIP dialog ‘800B0EEA-F28F-ED11-8172-D91ACB17E8B1@192.168.5.46’ in 6400 ms (Method: INVITE)

<— SIP read from TLS:192.168.5.46:49804 —>

ACK sip:4931@10.30.150.3 SIP/2.0

Via: SIP/2.0/TLS 192.168.5.46:49804;branch=z9hG4bK800b0eeaf28fed118173d91acb17e8b1;rport

From: sip:a44c119eb1f6@10.30.150.3;tag=1241913889

To: sip:4931@10.30.150.3;tag=as5a8db6cc

Call-ID: 800B0EEA-F28F-ED11-8172-D91ACB17E8B1@192.168.5.46

CSeq: 1 ACK

Content-Length: 0

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

— (7 headers 0 lines) —

<— SIP read from TLS:192.168.5.46:49804 —>

INVITE sip:4931@10.30.150.3 SIP/2.0

Via: SIP/2.0/TLS 192.168.5.46:49804;branch=z9hG4bK800b0eeaf28fed118174d91acb17e8b1;rport

From: sip:a44c119eb1f6@10.30.150.3;tag=1241913889

To: sip:4931@10.30.150.3

Call-ID: 800B0EEA-F28F-ED11-8172-D91ACB17E8B1@192.168.5.46

CSeq: 2 INVITE

Contact: sip:a44c119eb1f6@192.168.5.46:49804;transport=tls;gr=802F69CB-748F-ED11-806C-D91ACB17E8B1

Authorization: Digest username=“a44c119eb1f6”, realm=“asterisk”, nonce=“40c42f29”, uri="sip:4931@10.30.150.3", response=“34da4370fd134ca1ad9135fb579f6821”, algorithm=MD5

Content-Type: application/sdp

Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK

Max-Forwards: 70

Supported: 100rel, replaces, from-change, gruu

User-Agent: phoner 3.23

P-Preferred-Identity: sip:a44c119eb1f6@10.30.150.3

Content-Length: 676

v=0

o=- 663715420 1 IN IP4 192.168.5.46

s=phoner 3.23

c=IN IP4 192.168.5.46

t=0 0

m=audio 5063 RTP/AVP 8 0 2 3 97 9 18 111 112 113 114 11 118 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:3 GSM/8000

a=rtpmap:97 iLBC/8000

a=rtpmap:9 G722/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=yes

a=rtpmap:111 speex/16000

a=rtpmap:112 G726-16/8000

a=rtpmap:113 G726-24/8000

a=rtpmap:114 G726-40/8000

a=rtpmap:11 L16/44100

a=rtpmap:118 L16/16000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:M6ZTEkROmlEgl4EI92wCMlMhps/MiPI/w510EiUf

a=encryption:optional

a=ssrc:1103747463

a=sendrecv

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

— (15 headers 26 lines) —

Sending to 192.168.5.46:49804 (NAT)

Using INVITE request as basis request - 800B0EEA-F28F-ED11-8172-D91ACB17E8B1@192.168.5.46

Found peer ‘a44c119eb1f6’ for ‘a44c119eb1f6’ from 192.168.5.46:49804

== Using SIP VIDEO CoS mark 6

== Using SIP RTP CoS mark 5

Found RTP audio format 8

Found RTP audio format 0

Found RTP audio format 2

Found RTP audio format 3

Found RTP audio format 97

Found RTP audio format 9

Found RTP audio format 18

Found RTP audio format 111

Found RTP audio format 112

Found RTP audio format 113

Found RTP audio format 114

Found RTP audio format 11

Found RTP audio format 118

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 G726-32 for ID 2

Found audio description format GSM for ID 3

Found audio description format iLBC for ID 97

Found audio description format G722 for ID 9

Found audio description format G729 for ID 18

Found audio description format speex for ID 111

Found unknown media description format G726-16 for ID 112

Found unknown media description format G726-24 for ID 113

Found unknown media description format G726-40 for ID 114

Found unknown media description format L16 for ID 11

Found audio description format L16 for ID 118

Found audio description format telephone-event for ID 101

[Jan 11 08:56:17] NOTICE[11707][C-0000005c]: chan_sip.c:10742 process_sdp: Processed audio crypto attribute without SAVP specified; accepting anyway

Capabilities: us - (alaw|ulaw|gsm|g729|h263p|h263|h264|h261|vp8|vp9), peer - audio=(ulaw|gsm|alaw|g722|g729|ilbc|speex16|slin16)/video=(nothing)/text=(nothing), combined - (alaw|ulaw|gsm|g729)

Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

0x7fb03002f710 – Strict RTP learning after remote address set to: 192.168.5.46:5063

Peer audio RTP is at port 192.168.5.46:5063

Peer doesn’t provide video

Looking for 4931 in KosoveMobil (domain 10.30.150.3)

sip_route_dump: route/path hop: sip:a44c119eb1f6@192.168.5.46:49804;transport=tls;gr=802F69CB-748F-ED11-806C-D91ACB17E8B1

<— Transmitting (NAT) to 192.168.5.46:49804 —>

SIP/2.0 100 Trying

Via: SIP/2.0/TLS 192.168.5.46:49804;branch=z9hG4bK800b0eeaf28fed118174d91acb17e8b1;received=192.168.5.46;rport=49804

From: sip:a44c119eb1f6@10.30.150.3;tag=1241913889

To: sip:4931@10.30.150.3

Call-ID: 800B0EEA-F28F-ED11-8172-D91ACB17E8B1@192.168.5.46

CSeq: 2 INVITE

Server: Asterisk PBX 15.4.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Contact: sip:4931@10.30.150.3:5061;transport=tls

Content-Length: 0

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

– Executing [4931@KosoveMobil:1] Macro(“SIP/a44c119eb1f6-000000ac”, “stduser,SIP/00a2895ebf3b,30”) in new stack

– Executing [s@macro-stduser:1] NoOp(“SIP/a44c119eb1f6-000000ac”, ““EkoF Encrypted Line1” <4930> eshte duke thirrur 4931 ne SIP/00a2895ebf3b me 30 sekonda kohe cingerrime.”) in new stack

– Executing [s@macro-stduser:2] Dial(“SIP/a44c119eb1f6-000000ac”, “SIP/00a2895ebf3b,30”) in new stack

== Using SIP VIDEO CoS mark 6

== Using SIP RTP CoS mark 5

Audio is at 37004

Adding codec alaw to SDP

Adding codec ulaw to SDP

Adding codec gsm to SDP

Adding codec g729 to SDP

Adding non-codec 0x1 (telephone-event) to SDP

Reliably Transmitting (NAT) to 10.30.150.139:5071:

INVITE sip:00a2895ebf3b@10.30.150.139:5071;transport=tls SIP/2.0

Via: SIP/2.0/TLS 10.30.150.3:5061;branch=z9hG4bK22fdf963;rport

Max-Forwards: 70

From: “EkoF Encrypted Line1” sip:4930@10.30.150.3;tag=as2edd9dc6

To: sip:00a2895ebf3b@10.30.150.139:5071;transport=tls

Contact: sip:4930@10.30.150.3:5061;transport=tls

Call-ID: 4fd4e4954a57b82c5debfd9e1c8c8d39@10.30.150.3:5061

CSeq: 102 INVITE

User-Agent: Asterisk PBX 15.4.1

Date: Wed, 11 Jan 2023 07:56:17 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 416

v=0

o=root 1979425312 1979425312 IN IP4 10.30.150.3

s=Asterisk PBX 15.4.1

c=IN IP4 10.30.150.3

t=0 0

m=audio 37004 RTP/SAVP 8 0 3 18 101

a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:fYDWKUDXYL30aGw+SEWSSPTBc9t6Jpu8wBKXPRwA

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=maxptime:150

a=sendrecv


– Called SIP/00a2895ebf3b

<— SIP read from TLS:10.30.150.139:5071 —>

SIP/2.0 100 Trying

To: sip:00a2895ebf3b@10.30.150.139:5071

From: “EkoF Encrypted Line1” sip:4930@10.30.150.3;tag=as2edd9dc6

Call-ID: 4fd4e4954a57b82c5debfd9e1c8c8d39@10.30.150.3:5061

CSeq: 102 INVITE

Via: SIP/2.0/TLS 10.30.150.3:5061;branch=z9hG4bK22fdf963

Server: Cisco/SPA303-7.5.2

Content-Length: 0

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

— (8 headers 0 lines) —

<— SIP read from TLS:10.30.150.139:5071 —>

SIP/2.0 180 Ringing

To: sip:00a2895ebf3b@10.30.150.139:5071;tag=4106b848b3b11b02i0

From: “EkoF Encrypted Line1” sip:4930@10.30.150.3;tag=as2edd9dc6

Call-ID: 4fd4e4954a57b82c5debfd9e1c8c8d39@10.30.150.3:5061

CSeq: 102 INVITE

Via: SIP/2.0/TLS 10.30.150.3:5061;branch=z9hG4bK22fdf963

Contact: “EkoF 2 Encrypted Line2” sip:00a2895ebf3b@10.30.150.139:5071;transport=tls

Server: Cisco/SPA303-7.5.2

Content-Length: 0

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

— (9 headers 0 lines) —

sip_route_dump: route/path hop: sip:00a2895ebf3b@10.30.150.139:5071;transport=tls

– SIP/00a2895ebf3b-000000ad is ringing

<— Transmitting (NAT) to 192.168.5.46:49804 —>

SIP/2.0 180 Ringing

Via: SIP/2.0/TLS 192.168.5.46:49804;branch=z9hG4bK800b0eeaf28fed118174d91acb17e8b1;received=192.168.5.46;rport=49804

From: sip:a44c119eb1f6@10.30.150.3;tag=1241913889

To: sip:4931@10.30.150.3;tag=as6fe6af05

Call-ID: 800B0EEA-F28F-ED11-8172-D91ACB17E8B1@192.168.5.46

CSeq: 2 INVITE

Server: Asterisk PBX 15.4.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Contact: sip:4931@10.30.150.3:5061;transport=tls

Content-Length: 0

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

<— SIP read from TLS:10.30.150.139:5071 —>

SIP/2.0 200 OK

To: sip:00a2895ebf3b@10.30.150.139:5071;tag=4106b848b3b11b02i0

From: “EkoF Encrypted Line1” sip:4930@10.30.150.3;tag=as2edd9dc6

Call-ID: 4fd4e4954a57b82c5debfd9e1c8c8d39@10.30.150.3:5061

CSeq: 102 INVITE

Via: SIP/2.0/TLS 10.30.150.3:5061;branch=z9hG4bK22fdf963

Contact: “EkoF 2 Encrypted Line2” sip:00a2895ebf3b@10.30.150.139:5071;transport=tls

Server: Cisco/SPA303-7.5.2

Content-Length: 291

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE

Supported: replaces

Content-Type: application/sdp

v=0

o=- 33835 33835 IN IP4 10.30.150.139

s=-

c=IN IP4 10.30.150.139

t=0 0

m=audio 16440 RTP/SAVP 8 101

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv

a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:9n5UWvdO3hj6aq9yO2x94C5gUkq+63jorFJS4jeD

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

— (12 headers 12 lines) —

Found RTP audio format 8

Found RTP audio format 101

Found audio description format PCMA for ID 8

Found audio description format telephone-event for ID 101

Capabilities: us - (alaw|ulaw|gsm|g729|h263p|h263|h264|h261|vp8|vp9), 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|)

0x7fb014008570 – Strict RTP learning after remote address set to: 10.30.150.139:16440

Peer audio RTP is at port 10.30.150.139:16440

sip_route_dump: route/path hop: sip:00a2895ebf3b@10.30.150.139:5071;transport=tls

Transmitting (NAT) to 10.30.150.139:5071:

ACK sip:00a2895ebf3b@10.30.150.139:5071;transport=tls SIP/2.0

Via: SIP/2.0/TLS 10.30.150.3:5061;branch=z9hG4bK6d414d6b;rport

Max-Forwards: 70

From: “EkoF Encrypted Line1” sip:4930@10.30.150.3;tag=as2edd9dc6

To: sip:00a2895ebf3b@10.30.150.139:5071;transport=tls;tag=4106b848b3b11b02i0

Contact: sip:4930@10.30.150.3:5061;transport=tls

Call-ID: 4fd4e4954a57b82c5debfd9e1c8c8d39@10.30.150.3:5061

CSeq: 102 ACK

User-Agent: Asterisk PBX 15.4.1

Content-Length: 0


– SIP/00a2895ebf3b-000000ad answered SIP/a44c119eb1f6-000000ac

Audio is at 32504

Adding codec alaw to SDP

Adding codec ulaw to SDP

Adding codec gsm to SDP

Adding codec g729 to SDP

Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 192.168.5.46:49804 —>

SIP/2.0 200 OK

Via: SIP/2.0/TLS 192.168.5.46:49804;branch=z9hG4bK800b0eeaf28fed118174d91acb17e8b1;received=192.168.5.46;rport=49804

From: sip:a44c119eb1f6@10.30.150.3;tag=1241913889

To: sip:4931@10.30.150.3;tag=as6fe6af05

Call-ID: 800B0EEA-F28F-ED11-8172-D91ACB17E8B1@192.168.5.46

CSeq: 2 INVITE

Server: Asterisk PBX 15.4.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Contact: sip:4931@10.30.150.3:5061;transport=tls

Content-Type: application/sdp

Content-Length: 414

v=0

o=root 991565780 991565780 IN IP4 10.30.150.3

s=Asterisk PBX 15.4.1

c=IN IP4 10.30.150.3

t=0 0

m=audio 32504 RTP/SAVP 8 0 3 18 101

a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:jEN5r2rUjDaajNuTXjJDI+mZooewz7ibzNObI7lN

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=maxptime:150

a=sendrecv

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

<— SIP read from TLS:192.168.5.46:49804 —>

ACK sip:4931@10.30.150.3:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.5.46:49804;branch=z9hG4bK00cfd7ebf28fed118174d91acb17e8b1;rport

From: sip:a44c119eb1f6@10.30.150.3;tag=1241913889

To: sip:4931@10.30.150.3;tag=as6fe6af05

Call-ID: 800B0EEA-F28F-ED11-8172-D91ACB17E8B1@192.168.5.46

CSeq: 2 ACK

Contact: sip:a44c119eb1f6@192.168.5.46:49804;transport=tls;gr=802F69CB-748F-ED11-806C-D91ACB17E8B1

Authorization: Digest username=“a44c119eb1f6”, realm=“asterisk”, nonce=“40c42f29”, uri=“sip:4931@10.30.150.3:5061;transport=tls”, response=“597e24efe121962c39461d6895c54e31”, algorithm=MD5

Max-Forwards: 70

Content-Length: 0

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

— (10 headers 0 lines) —

– Channel SIP/00a2895ebf3b-000000ad joined ‘simple_bridge’ basic-bridge <9c567c0c-4e27-4e51-bc7e-e046c3ad447e>

– Channel SIP/a44c119eb1f6-000000ac joined ‘simple_bridge’ basic-bridge <9c567c0c-4e27-4e51-bc7e-e046c3ad447e>

0x7fb03002f710 – Strict RTP switching to RTP target address 192.168.5.46:5063 as source

Got RTP packet from 192.168.5.46:5063 (type 08, seq 000001, ts 006210, len 000160)

Got RTP packet from 192.168.5.46:5063 (type 08, seq 000002, ts 006370, len 000160)

Sent RTP packet to 10.30.150.139:16440 (type 08, seq 029581, ts 000088, len 000250)

Got RTP packet from 192.168.5.46:5063 (type 08, seq 000003, ts 006530, len 000160)

Sent RTP packet to 10.30.150.139:16440 (type 08, seq 029582, ts 000328, len 000250)

Got RTP packet from 192.168.5.46:5063 (type 08, seq 000004, ts 006690, len 000160)

0x7fb014008570 – Strict RTP switching to RTP target address 10.30.150.139:16440 as source

Got RTP packet from 10.30.150.139:16440 (type 08, seq 006328, ts 196179362, len 000240)

Sent RTP packet to 192.168.5.46:5063 (type 08, seq 001702, ts 196179360, len 000250)

Got RTP packet from 192.168.5.46:5063 (type 08, seq 000005, ts 006850, len 000160)

Sent RTP packet to 10.30.150.139:16440 (type 08, seq 029583, ts 000568, len 000250)

Got RTP packet from 192.168.5.46:5063 (type 08, seq 000006, ts 007010, len 000160)

Sent RTP packet to 10.30.150.139:16440 (type 08, seq 029584, ts 000808, len 000250)

Got RTP packet from 10.30.150.139:16440 (type 08, seq 006329, ts 196179602, len 000240)

Sent RTP packet to 192.168.5.46:5063 (type 08, seq 001703, ts 196179600, len 000250)

Got RTP packet from 192.168.5.46:5063 (type 08, seq 000007, ts 007170, len 000160)

Got RTP packet from 10.30.150.139:16440 (type 08, seq 006330, ts 196179842, len 000240)

Sent RTP packet to 192.168.5.46:5063 (type 08, seq 001704, ts 196179840, len 000250)

Got RTP packet from 192.168.5.46:5063 (type 08, seq 000008, ts 007330, len 000160)

Sent RTP packet to 10.30.150.139:16440 (type 08, seq 029585, ts 001048, len 000250)

Got RTP packet from 192.168.5.46:5063 (type 08, seq 000009, ts 007490, len 000160)

Sent RTP packet to 10.30.150.139:16440 (type 08, seq 029586, ts 001288, len 000250)

Got RTP packet from 10.30.150.139:16440 (type 08, seq 006331, ts 196180082, len 000240)

Sent RTP packet to 192.168.5.46:5063 (type 08, seq 001705, ts 196180080, len 000250)

Got RTP packet from 192.168.5.46:5063 (type 08, seq 000010, ts 007650, len 000160)

Got RTP packet from 10.30.150.139:16440 (type 08, seq 006332, ts 196180322, len 000240)

Sent RTP packet to 192.168.5.46:5063 (type 08, seq 001706, ts 196180320, len 000250)

Got RTP packet from 192.168.5.46:5063 (type 08, seq 000011, ts 007810, len 000160)

Sent RTP packet to 10.30.150.139:16440 (type 08, seq 029587, ts 001528, len 000250)

Got RTP packet from 192.168.5.46:5063 (type 08, seq 000012, ts 007970, len 000160)

Sent RTP packet to 10.30.150.139:16440 (type 08, seq 029588, ts 001768, len 000250)

Got RTP packet from 10.30.150.139:16440 (type 08, seq 006333, ts 196180562, len 000240)

RTP Packets sent/received shown above on the dialog but on the phone web showing 0 - Zero received:

<— SIP read from TLS:10.30.150.139:5071 —>

BYE sip:4930@10.30.150.3:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 10.30.150.139:5071;branch=z9hG4bK-ccf172e1

From: sip:00a2895ebf3b@10.30.150.139;transport=tls;tag=4106b848b3b11b02i0

To: “EkoF Encrypted Line1” sip:4930@10.30.150.3;tag=as2edd9dc6

Call-ID: 4fd4e4954a57b82c5debfd9e1c8c8d39@10.30.150.3:5061

CSeq: 101 BYE

Max-Forwards: 70

User-Agent: Cisco/SPA303-7.5.2

Content-Length: 0

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

— (9 headers 0 lines) —

Sending to 10.30.150.139:5071 (NAT)

Got RTP packet from 192.168.5.46:5063 (type 08, seq 000061, ts 015810, len 000160)

Scheduling destruction of SIP dialog ‘4fd4e4954a57b82c5debfd9e1c8c8d39@10.30.150.3:5061’ in 6400 ms (Method: BYE)

<— Transmitting (NAT) to 10.30.150.139:5071 —>

SIP/2.0 200 OK

Via: SIP/2.0/TLS 10.30.150.139:5071;branch=z9hG4bK-ccf172e1;received=10.30.150.139;rport=5071

From: sip:00a2895ebf3b@10.30.150.139;transport=tls;tag=4106b848b3b11b02i0

To: “EkoF Encrypted Line1” sip:4930@10.30.150.3;tag=as2edd9dc6

Call-ID: 4fd4e4954a57b82c5debfd9e1c8c8d39@10.30.150.3:5061

CSeq: 101 BYE

Server: Asterisk PBX 15.4.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Length: 0

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

Got RTP packet from 192.168.5.46:5063 (type 08, seq 000062, ts 015970, len 000160)

– Channel SIP/00a2895ebf3b-000000ad left ‘simple_bridge’ basic-bridge <9c567c0c-4e27-4e51-bc7e-e046c3ad447e>

– Channel SIP/a44c119eb1f6-000000ac left ‘simple_bridge’ basic-bridge <9c567c0c-4e27-4e51-bc7e-e046c3ad447e>

== Spawn extension (macro-stduser, s, 2) exited non-zero on ‘SIP/a44c119eb1f6-000000ac’ in macro ‘stduser’

== Spawn extension (KosoveMobil, 4931, 1) exited non-zero on ‘SIP/a44c119eb1f6-000000ac’

Scheduling destruction of SIP dialog ‘800B0EEA-F28F-ED11-8172-D91ACB17E8B1@192.168.5.46’ in 6400 ms (Method: ACK)

Reliably Transmitting (NAT) to 192.168.5.46:49804:

BYE sip:a44c119eb1f6@192.168.5.46:49804;transport=tls;gr=802F69CB-748F-ED11-806C-D91ACB17E8B1 SIP/2.0

Via: SIP/2.0/TLS 10.30.150.3:5061;branch=z9hG4bK68d11c5e;rport

Max-Forwards: 70

From: sip:4931@10.30.150.3;tag=as6fe6af05

To: sip:a44c119eb1f6@10.30.150.3;tag=1241913889

Call-ID: 800B0EEA-F28F-ED11-8172-D91ACB17E8B1@192.168.5.46

CSeq: 102 BYE

User-Agent: Asterisk PBX 15.4.1

Proxy-Authorization: Digest username=“4930”, realm=“asterisk”, algorithm=MD5, uri=“sips:10.30.150.3”, nonce=“40c42f29”, response=“1d468248427a184eb9c5e745e471d983”

X-Asterisk-HangupCause: Normal Clearing

X-Asterisk-HangupCauseCode: 16

Content-Length: 0


<— SIP read from TLS:192.168.5.46:49804 —>

SIP/2.0 200 OK

Via: SIP/2.0/TLS 10.30.150.3:5061;branch=z9hG4bK68d11c5e;rport=5061

From: sip:4931@10.30.150.3;tag=as6fe6af05

To: sip:a44c119eb1f6@10.30.150.3;tag=1241913889

Call-ID: 800B0EEA-F28F-ED11-8172-D91ACB17E8B1@192.168.5.46

CSeq: 102 BYE

Contact: sip:a44c119eb1f6@192.168.5.46:5062;transport=tls;gr=802F69CB-748F-ED11-806C-D91ACB17E8B1

Server: phoner 3.23

Content-Length: 0

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

— (9 headers 0 lines) —

SIP Response message for INCOMING dialog BYE arrived

Really destroying SIP dialog ‘800B0EEA-F28F-ED11-8172-D91ACB17E8B1@192.168.5.46’ Method: ACK

Really destroying SIP dialog ‘cdb36720-6903fbf9@10.30.150.74’ Method: REGISTER

Reliably Transmitting (NAT) to 10.30.150.139:5071:

OPTIONS sip:00a2895ebf3b@10.30.150.139:5071;transport=tls SIP/2.0

Via: SIP/2.0/TLS 10.30.150.3:5061;branch=z9hG4bK2fcbcfed;rport

Max-Forwards: 70

From: “asterisk” sip:asterisk@10.30.150.3;tag=as5841583c

To: sip:00a2895ebf3b@10.30.150.139:5071;transport=tls

Contact: sip:asterisk@10.30.150.3:5061;transport=tls

Call-ID: 46a62f874cbc003b2fe655886819acf7@10.30.150.3:5061

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX 15.4.1

Date: Wed, 11 Jan 2023 07:56:22 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Length: 0


<— SIP read from TLS:10.30.150.139:5071 —>

SIP/2.0 200 OK

To: sip:00a2895ebf3b@10.30.150.139:5071;tag=a131da9bd4522d01i0

From: “asterisk” sip:asterisk@10.30.150.3;tag=as5841583c

Call-ID: 46a62f874cbc003b2fe655886819acf7@10.30.150.3:5061

CSeq: 102 OPTIONS

Via: SIP/2.0/TLS 10.30.150.3:5061;branch=z9hG4bK2fcbcfed

Server: Cisco/SPA303-7.5.2

Content-Length: 0

Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE

Supported: replaces

kpvoip*CLI> sip show channel 8deaf05c-52589e@10.30.150.74

  • SIP Call
    Curr. trans. direction: Incoming
    Call-ID: 8deaf05c-52589e@10.30.150.74
    Owner channel ID:
    Our Codec Capability: (alaw|h263p|h263|h264|h261|vp8|vp9)
    Non-Codec Capability (DTMF): 1
    Their Codec Capability: (ulaw|g726|alaw|g722|g729)
    Joint Codec Capability: (alaw)
    Format: (nothing)
    T.38 support No
    Video support No
    MaxCallBR: 3000 kbps
    Theoretical Address: 10.30.150.74:5068
    Received Address: 10.30.150.74:5068
    SIP Transfer mode: open
    Force rport: Yes
    Audio IP: 10.30.150.3 (local)
    Our Tag: as44b4e279
    Their Tag: 10f3d44cc1c30d6o0
    SIP User agent: Cisco/SPA504G-7.5.2b
    Username: 4930
    Peername: a44c119eb1f6
    Original uri: sip:a44c119eb1f6@10.30.150.74:5068
    Caller-ID: 4930
    Need Destroy: No
    Last Message: Rx: INVITE
    Promiscuous Redir: No
    Route: sip:a44c119eb1f6@10.30.150.74:5068;transport=tls
    DTMF Mode: rfc2833
    SIP Options: replaces replace
    Session-Timer: Inactive
    Transport: TLS
    Media: SRTP

Subscription (type: mwi)
Curr. trans. direction: Outgoing
Call-ID: 802F69CB-748F-ED11-806B-D91ACB17E8B1@192.168.5.46
Owner channel ID:
Our Codec Capability: (alaw|ulaw|gsm|g729|h263p|h263|h264|h261|vp8|vp9)
Non-Codec Capability (DTMF): 1
Their Codec Capability: (nothing)
Joint Codec Capability: (alaw|ulaw|gsm|g729|h263p|h263|h264|h261|vp8|vp9)
Format: (nothing)
T.38 support No
Video support No
MaxCallBR: 3000 kbps
Theoretical Address: 192.168.5.46:49804
Received Address: 192.168.5.46:49804
SIP Transfer mode: open
Force rport: Yes
Audio IP: 10.30.150.3 (local)
Our Tag: as3b798dac
Their Tag: 3469055167
SIP User agent: phoner 3.23
Username: 4930
Peername: a44c119eb1f6
Original uri: sip:a44c119eb1f6@192.168.5.46:5062
Caller-ID: 4930
Need Destroy: No
Last Message: Init: NOTIFY
Promiscuous Redir: No
Route: sip:a44c119eb1f6@192.168.5.46:49804;transport=tls
DTMF Mode: rfc2833
SIP Options: (none)
Session-Timer: Inactive
Transport: TLS
Media: None