[Linphone-users] SRTP troubles between Asterisk and sip.linphone.org

[Andrej Mikus]

Team,

I am trying to dig deep into SIP and VoIP, and I think I have certain 
progress. My Linphone clients in desktops and Android call register 
happily to my Asterisk/Freepbx server and call between each other, and 
also with native Android SIP client or Cisco IP Phones.

I have setup an account at sip.linphone.org and made it a trunk in the 
PBX, using pjsip channel. This also works well with UDP, TCP, TLS, as 
long as I do not try to encrypt the call itself. As soon as I do that,
the Linphone server responds with SIP/2.0 488 Not acceptable here

I am failing to figure out what could be wrong in my client side config. 
Is there any documentation of the sip service I could check to figure 
out more? Ideally with use of asterisk and pjsip channel as a client?

Below is my session recorded with detail logging. Is the offer of 
RTP/SAVP explicitly rejected or my client is missing something?

I will be thankful for any pointer or hint.

Regards
Andrej Mikus

<--- Transmitting SIP request (1172 bytes) to TLS:54.37.202.229:5223 --->
INVITE sip:thetestcall@sip.linphone.org SIP/2.0
Via: SIP/2.0/TLS 
192.168.5.5:5061;rport;branch=z9hG4bKPj1d1b8532-dada-476e-8013-0e1ac9d53145;alias
From: 
<sip:my_account_id@sip.linphone.org>;tag=41661d8f-40da-4ffd-860e-cc677748893a
To: <sip:thetestcall@sip.linphone.org>
Contact: <sip:my_account_id@192.168.5.5:5061;transport=TLS>
Call-ID: ffc1cc2c-d6ea-49e3-8194-da228e619aa8
CSeq: 2281 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, 
CANCEL, UPDATE, PRACK, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u2
Content-Type: application/sdp
Content-Length:   469

v=0
o=- 1731182403 1731182403 IN IP4 192.168.5.5
s=Asterisk
c=IN IP4 192.168.5.5
t=0 0
m=audio 12672 RTP/SAVP 0 8 18 3 111 9 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:+Jg5tuIPuxtZQ4xtMGLF+XHvV0EEuAhjc6WvySvi
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (592 bytes) from TLS:54.37.202.229:5223 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/TLS 
192.168.5.5:5061;rport=45281;branch=z9hG4bKPj1d1b8532-dada-476e-8013-0e1ac9d53145;alias;received=84.192.184.58
From: 
<sip:my_account_id@sip.linphone.org>;tag=41661d8f-40da-4ffd-860e-cc677748893a
To: <sip:thetestcall@sip.linphone.org>;tag=Nj3XQmK7t5y0D
Call-ID: ffc1cc2c-d6ea-49e3-8194-da228e619aa8
CSeq: 2281 INVITE
Server: Flexisip/2.0.3-9-g60ae233c (sofia-sip-nta/2.0)
Proxy-Authenticate: Digest realm="sip.linphone.org", 
nonce="LMbF4wAAAADIJv2mAAD0xo5BVewAAAAA", opaque="+GNywA==", 
algorithm=MD5, qop="auth"
Content-Length: 0


<--- Transmitting SIP request (433 bytes) to TLS:54.37.202.229:5223 --->
ACK sip:thetestcall@sip.linphone.org SIP/2.0
Via: SIP/2.0/TLS 
192.168.5.5:5061;rport;branch=z9hG4bKPj1d1b8532-dada-476e-8013-0e1ac9d53145;alias
From: 
<sip:my_account_id@sip.linphone.org>;tag=41661d8f-40da-4ffd-860e-cc677748893a
To: <sip:thetestcall@sip.linphone.org>;tag=Nj3XQmK7t5y0D
Call-ID: ffc1cc2c-d6ea-49e3-8194-da228e619aa8
CSeq: 2281 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u2
Content-Length:  0


<--- Transmitting SIP request (1476 bytes) to TLS:54.37.202.229:5223 --->
INVITE sip:thetestcall@sip.linphone.org SIP/2.0
Via: SIP/2.0/TLS 
192.168.5.5:5061;rport;branch=z9hG4bKPjfd76bc45-7c48-4ae9-9d3b-10742999e572;alias
From: 
<sip:my_account_id@sip.linphone.org>;tag=41661d8f-40da-4ffd-860e-cc677748893a
To: <sip:thetestcall@sip.linphone.org>
Contact: <sip:my_account_id@192.168.5.5:5061;transport=TLS>
Call-ID: ffc1cc2c-d6ea-49e3-8194-da228e619aa8
CSeq: 2282 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, 
CANCEL, UPDATE, PRACK, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u2
Proxy-Authorization: Digest username="my_account_id", 
realm="sip.linphone.org", nonce="LMbF4wAAAADIJv2mAAD0xo5BVewAAAAA", 
uri="sip:thetestcall@sip.linphone.org", 
response="b252c4e5f71f3609cc6706d5c0d49b2a", algorithm=MD5, 
cnonce="45300d44-f971-40d7-bd70-e24a8cc2cdcb", opaque="+GNywA==", 
qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   469

v=0
o=- 1731182403 1731182403 IN IP4 192.168.5.5
s=Asterisk
c=IN IP4 192.168.5.5
t=0 0
m=audio 12672 RTP/SAVP 0 8 18 3 111 9 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:+Jg5tuIPuxtZQ4xtMGLF+XHvV0EEuAhjc6WvySvi
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (458 bytes) from TLS:54.37.202.229:5223 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 
192.168.5.5:5061;rport=45281;branch=z9hG4bKPjfd76bc45-7c48-4ae9-9d3b-10742999e572;alias;received=84.192.184.58
Record-Route: <sips:sip6.linphone.org:5223;lr>
From: 
<sip:my_account_id@sip.linphone.org>;tag=41661d8f-40da-4ffd-860e-cc677748893a
To: <sip:thetestcall@sip.linphone.org>
Call-ID: ffc1cc2c-d6ea-49e3-8194-da228e619aa8
CSeq: 2282 INVITE
Server: Flexisip/2.0.3-9-g60ae233c (sofia-sip-nta/2.0)
Content-Length: 0


<--- Received SIP response (531 bytes) from TLS:54.37.202.229:5223 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/TLS 
192.168.5.5:5061;rport=45281;branch=z9hG4bKPjfd76bc45-7c48-4ae9-9d3b-10742999e572;alias;received=84.192.184.58
From: 
<sip:my_account_id@sip.linphone.org>;tag=41661d8f-40da-4ffd-860e-cc677748893a
To: <sip:thetestcall@sip.linphone.org>;tag=as0aa55d55
Call-ID: ffc1cc2c-d6ea-49e3-8194-da228e619aa8
CSeq: 2282 INVITE
Server: Asterisk PBX 16.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<--- Transmitting SIP request (430 bytes) to TLS:54.37.202.229:5223 --->
ACK sip:thetestcall@sip.linphone.org SIP/2.0
Via: SIP/2.0/TLS 
192.168.5.5:5061;rport;branch=z9hG4bKPjfd76bc45-7c48-4ae9-9d3b-10742999e572;alias
From: 
<sip:my_account_id@sip.linphone.org>;tag=41661d8f-40da-4ffd-860e-cc677748893a
To: <sip:thetestcall@sip.linphone.org>;tag=as0aa55d55
Call-ID: ffc1cc2c-d6ea-49e3-8194-da228e619aa8
CSeq: 2282 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u2
Content-Length:  0


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