488 Not acceptable here on FreePBX

[chyiannakou]

Hi,

I registered two extensions in Chrome and Firefox but cannot establish a call from any extension or to any of these extensions. I get the error "488 Not acceptable here".
Both Asterisk and SIPML5 are on the same machine
I use Freepbx with Asterisk 11.13.1.

Find below the SIP debugging log files for your review.

Please advice

<--- SIP read from WS:10.111.31.244:49473 --->
INVITE sip:80...@10.112.155.200 SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=

z9hG4bKqh4v7k1G7aTaUpi2uoeqozABuU8mpOQw;rport
From: "webRTC"<sip:80...@10.112.155.200>;tag=TGPoXiDc7quXfDb3SfXV
To: <sip:80...@10.112.155.200>
Contact: "webRTC"<sips:8001@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=wss>;impi=8001;ha1=60ef98abeb1fec15636f4735a544349d;+g.oma.sip-im;language="en,fr"
Call-ID: c987050a-230a-ce6c-f432-f640449b1025
CSeq: 23880 INVITE
Content-Type: application/sdp
Content-Length: 1016
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

v=0
o=Mozilla-SIPUA-35.0.1 6610 0 IN IP4 0.0.0.0
s=Doubango Telecom - firefox
t=0 0
a=ice-ufrag:f0a61510
a=ice-pwd:112cf60a40728392097aafd288168ee2
a=fingerprint:sha-256 91:A1:7F:C1:B7:04:41:65:6E:B7:88:3F:F9:CA:5A:61:B3:B2:B0:2C:91:23:16:8A:E1:E0:E9:C8:89:CD:BD:43
m=audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 0.0.0.0
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=setup:actpass
a=rtcp-mux
a=candidate:0 1 UDP 2128609535 10.111.31.244 56102 typ host
a=candidate:2 1 UDP 2128543999 192.168.0.126 56103 typ host
a=candidate:0 2 UDP 2128609534 10.111.31.244 56104 typ host
a=candidate:2 2 UDP 2128543998 192.168.0.126 56105 typ host
a=candidate:1 1 UDP 1692467199 85.207.0.55 56102 typ srflx raddr 10.111.31.244 rport 56102
a=candidate:1 2 UDP 1692467198 85.207.0.55 56104 typ srflx raddr 10.111.31.244 rport 56104
<------------->
--- (12 headers 26 lines) ---
Using INVITE request as basis request - c987050a-230a-ce6c-f432-f640449b1025
Found peer '8001' for '8001' from 10.111.31.244:49473

<--- Reliably Transmitting (no NAT) to 10.111.31.244:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKqh4v7k1G7aTaUpi2uoeqozABuU8mpOQw;rport;received=10.111.31.244
From: "webRTC"<sip:80...@10.112.155.200>;tag=TGPoXiDc7quXfDb3SfXV
To: <sip:80...@10.112.155.200>;tag=as26041df2
Call-ID: c987050a-230a-ce6c-f432-f640449b1025
CSeq: 23880 INVITE
Server: FPBX-2.11.0(11.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="10.112.155.20", nonce="03f6ed88"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'c987050a-230a-ce6c-f432-f640449b1025' in 9152 ms (Method: INVITE)
  == Manager 'admin' logged off from 127.0.0.1

<--- SIP read from WS:10.111.31.244:49473 --->
ACK sip:80...@10.112.155.200 SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKqh4v7k1G7aTaUpi2uoeqozABuU8mpOQw;rport
From: "webRTC"<sip:80...@10.112.155.200>;tag=TGPoXiDc7quXfDb3SfXV
To: <sip:80...@10.112.155.200>;tag=as26041df2
Call-ID: c987050a-230a-ce6c-f432-f640449b1025
CSeq: 23880 ACK
Content-Length: 0
Max-Forwards: 70

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

<--- SIP read from WS:10.111.31.244:49473 --->
INVITE sip:80...@10.112.155.200 SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKg9LEmu2PRPaeBrnoAe2gO3DiDRQX1hgT;rport
From: "webRTC"<sip:80...@10.112.155.200>;tag=TGPoXiDc7quXfDb3SfXV
To: <sip:80...@10.112.155.200>
Contact: "webRTC"<sips:8001@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=wss>;impi=8001;ha1=60ef98abeb1fec15636f4735a544349d;+g.oma.sip-im;language="en,fr"
Call-ID: c987050a-230a-ce6c-f432-f640449b1025
CSeq: 23881 INVITE
Content-Type: application/sdp
Content-Length: 1016
Max-Forwards: 70
Authorization: Digest username="8001",realm="10.112.155.20",nonce="03f6ed88",uri="sip:80...@10.112.155.200",response="0f641e1130bf7bdfcf507243aa9e1221",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

v=0
o=Mozilla-SIPUA-35.0.1 6610 0 IN IP4 0.0.0.0
s=Doubango Telecom - firefox
t=0 0
a=ice-ufrag:f0a61510
a=ice-pwd:112cf60a40728392097aafd288168ee2
a=fingerprint:sha-256 91:A1:7F:C1:B7:04:41:65:6E:B7:88:3F:F9:CA:5A:61:B3:B2:B0:2C:91:23:16:8A:E1:E0:E9:C8:89:CD:BD:43
m=audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 0.0.0.0
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=setup:actpass
a=rtcp-mux
a=candidate:0 1 UDP 2128609535 10.111.31.244 56102 typ host
a=candidate:2 1 UDP 2128543999 192.168.0.126 56103 typ host
a=candidate:0 2 UDP 2128609534 10.111.31.244 56104 typ host
a=candidate:2 2 UDP 2128543998 192.168.0.126 56105 typ host
a=candidate:1 1 UDP 1692467199 85.207.0.55 56102 typ srflx raddr 10.111.31.244 rport 56102
a=candidate:1 2 UDP 1692467198 85.207.0.55 56104 typ srflx raddr 10.111.31.244 rport 56104
<------------->
--- (13 headers 26 lines) ---
Using INVITE request as basis request - c987050a-230a-ce6c-f432-f640449b1025
Found peer '8001' for '8001' from 10.111.31.244:49473
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
Found RTP audio format 109
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found unknown media description format opus for ID 109
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
[2015-02-09 21:53:25] WARNING[54377][C-0000003e]: chan_sip.c:10384 process_sdp: Rejecting secure audio stream without encryption details: audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8 101

<--- Reliably Transmitting (no NAT) to 10.111.31.244:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKg9LEmu2PRPaeBrnoAe2gO3DiDRQX1hgT;rport;received=10.111.31.244
From: "webRTC"<sip:80...@10.112.155.200>;tag=TGPoXiDc7quXfDb3SfXV
To: <sip:80...@10.112.155.200>;tag=as26041df2
Call-ID: c987050a-230a-ce6c-f432-f640449b1025
CSeq: 23881 INVITE
Server: FPBX-2.11.0(11.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

[navaismo]

Configure your FreePBX box to use DTLS-SRTP.

There are already many tutorials to do that, and in firefox dont use selfsigned certificates.

El lunes, 16 de febrero de 2015, 1:38:53 (UTC-6), chyiannakou escribió:

Hi,

I registered two extensions in Chrome and Firefox but cannot establish a call from any extension or to any of these extensions. I get the error "488 Not acceptable here".
Both Asterisk and SIPML5 are on the same machine
I use Freepbx with Asterisk 11.13.1.

Find below the SIP debugging log files for your review.

Please advice

<--- SIP read from WS:10.111.31.244:49473 --->

INVITE sip:...@10.112.155.200 SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=

z9hG4bKqh4v7k1G7aTaUpi2uoeqozABuU8mpOQw;rport
From: "webRTC"<sip:...@10.112.155.200>;tag=TGPoXiDc7quXfDb3SfXV
To: <sip:...@10.112.155.200>

From: "webRTC"<sip:...@10.112.155.200>;tag=TGPoXiDc7quXfDb3SfXV
To: <sip:...@10.112.155.200>;tag=as26041df2

Call-ID: c987050a-230a-ce6c-f432-f640449b1025
CSeq: 23880 INVITE
Server: FPBX-2.11.0(11.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="10.112.155.20", nonce="03f6ed88"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'c987050a-230a-ce6c-f432-f640449b1025' in 9152 ms (Method: INVITE)
  == Manager 'admin' logged off from 127.0.0.1

<--- SIP read from WS:10.111.31.244:49473 --->

ACK sip:...@10.112.155.200 SIP/2.0

Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKqh4v7k1G7aTaUpi2uoeqozABuU8mpOQw;rport

From: "webRTC"<sip:...@10.112.155.200>;tag=TGPoXiDc7quXfDb3SfXV
To: <sip:...@10.112.155.200>;tag=as26041df2

Call-ID: c987050a-230a-ce6c-f432-f640449b1025
CSeq: 23880 ACK
Content-Length: 0
Max-Forwards: 70

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

<--- SIP read from WS:10.111.31.244:49473 --->

INVITE sip:...@10.112.155.200 SIP/2.0

Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKg9LEmu2PRPaeBrnoAe2gO3DiDRQX1hgT;rport

From: "webRTC"<sip:...@10.112.155.200>;tag=TGPoXiDc7quXfDb3SfXV
To: <sip:...@10.112.155.200>

From: "webRTC"<sip:...@10.112.155.200>;tag=TGPoXiDc7quXfDb3SfXV
To: <sip:...@10.112.155.200>;tag=as26041df2

[christodoulos yiannakou]

Hi,

I configured my FreePBX to use SRTP. I tested the outgoing calls from extensions that use as media the SRTP protocol.

But again when it is about the SIPML5 extension i cannot perform calls. I still get the same error message
"WARNING[30289][C-0000006c]: chan_sip.c:10384 process_sdp: Rejecting secure audio stream without encryption details: audio 55702 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126"

The SIPML5 extension configuration has

avpf =yes

icesupport=yes

transport = All- WS primary

encryption = YES (STRP only)

Do i need any other configuration on the SIPML5 extension? Can you advice what i should look for?

Thank you

Chris

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Regards,


Christodoulos

[chyiannakou]

[navaismo]

You need to configure it for DTLS-SRTP not only for use srtp. Your issue was explained a lot here and in the asterisk forums. You need to create certificates for your peers and enable the dtls support on each peer.

[chyiannakou]

Hi

I did that also. I searched through this forum and it is a mess. I saw other posts that experienced the same issue with me without any solution.
I also followed your guide https://code.google.com/p/sipml5/wiki/Asterisk but again the same.

Is there any guide that will tell you what steps you should follow and in order to implement this configuration? you mention there is a lot of documentation but personally i did not find any. I would appreciate if you send me a related article.

Thanks.

Chris

[navaismo]

Here: http://forums.digium.com/viewtopic.php?f=1&t=91007&sid=56b281adf99707ba614447b4e83a9399

[chyiannakou]

Hi again,

Following the https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 for the prerequisites it mentions "Using menuselect make sure Asterisk will build with res_http_websocket, res_crypto and chan_sip."

During the compilation of Asterisk these packages are included. I did the configuration but i keep taking the message
"[2015-03-03 18:31:26] WARNING[14160][C-00000006]: chan_sip.c:10384 process_sdp: Rejecting secure audio stream without encryption details: audio 56387 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126"

Can you advice?

Chris

[navaismo]

If you are using FreePBX try updating your system freepbx is compatible with webrtc, also check your build with the command:

rpm -qa | grep uuid && echo && ldd /usr/lib/asterisk/modules/res_rtp_asterisk.so && echo && ls -lha /lib/libuu* && cat /etc/issue

replace lib with lib64 if apply.

[chyiannakou]

Hi Navaismo,

I upgraded my FreePBX with Asterisk 11.16.0 that works for me in a debian setup, but again i get the same error.

When i run the command rpm -qa | grep uuid && echo && ldd /usr/lib/asterisk/modules/res_rtp_asterisk.so && echo && ls -lha /lib/libuu* && cat /etc/issue i get the following message:
"warning: Generating 12 missing index(es), please wait..."

Can you let me know what does this means?

Thank you

Christodoulos

[navaismo]

That command is for CentOS. In your system locate the asterisk's module dir and then run:

ldd res_rtp_asterisk.so

In your other mail you said that your system was working so is working or not?

[chyiannakou]

I have two systems. One Freepbx running Asterisk 11.13.1 installed in slackware linux and one running FreePBX with Asterisk 11.13.1 in debian linux. The one in Debian linux after upgrading Asterisk 11 to the latest distribution (11.16.0) it works.

The slackware system although upgraded also to Asterisk 11.16.0 still show this error message.
 
# ldd res_rtp_asterisk.so
        linux-vdso.so.1 (0x00007fff821ff000)
        libuuid.so.1 => /lib64/libuuid.so.1 (0x00007fe9e8006000)
        libpthread.so.0 => /lib64/libpthread.so.0 (0x00007fe9e7de9000)
        libc.so.6 => /lib64/libc.so.6 (0x00007fe9e7a1e000)
        /lib64/ld-linux-x86-64.so.2 (0x00007fe9e843f000)

The output in the Debian working system is the following:

s#  ldd res_rtp_asterisk.so
        linux-gate.so.1 =>  (0xb7741000)
        libavformat.so.54 => /usr/local/lib/libavformat.so.54 (0xb76ad000)
        libavcodec.so.54 => /usr/local/lib/libavcodec.so.54 (0xb72a3000)
        libswscale.so.2 => /usr/local/lib/libswscale.so.2 (0xb724e000)
        libavutil.so.51 => /usr/local/lib/libavutil.so.51 (0xb7221000)
        libuuid.so.1 => /lib/i386-linux-gnu/libuuid.so.1 (0xb721b000)
        libm.so.6 => /lib/i386-linux-gnu/i686/cmov/libm.so.6 (0xb71f5000)
        libnsl.so.1 => /lib/i386-linux-gnu/i686/cmov/libnsl.so.1 (0xb71de000)
        librt.so.1 => /lib/i386-linux-gnu/i686/cmov/librt.so.1 (0xb71d4000)
        libpthread.so.0 => /lib/i386-linux-gnu/i686/cmov/libpthread.so.0 (0xb71bb000)
        libcrypto.so.1.0.0 => /usr/lib/i386-linux-gnu/i686/cmov/libcrypto.so.1.0.0 (0xb6ffc000)
        libssl.so.1.0.0 => /usr/lib/i386-linux-gnu/i686/cmov/libssl.so.1.0.0 (0xb6fa2000)
        libc.so.6 => /lib/i386-linux-gnu/i686/cmov/libc.so.6 (0xb6e3e000)
        libx264.so.123 => /usr/lib/i386-linux-gnu/i686/sse2/libx264.so.123 (0xb6c9f000)
        /lib/ld-linux.so.2 (0xb7742000)
        libdl.so.2 => /lib/i386-linux-gnu/i686/cmov/libdl.so.2 (0xb6c9b000)
        libz.so.1 => /lib/i386-linux-gnu/libz.so.1 (0xb6c82000)


So i guess there are some packages that are missing in the slackware or am i wrong?

Christodoulos

[navaismo]

Try running the install_prereq script from asterisk crontib dir. Then run the make distclean, and recompile all.

[chyiannakou]

Already tried to run install_prereq script from asterisk crontib dir but i get the message that this command is not supported by my distribution.

#./install_prereq install
./install_prereq: Your distribution (Slackware) is currently not supported. Aborting.

Is there any other way to find and install all the prerequisites from tarballs?

Christodoulos

[navaismo]

Ah right that only works for Redhat/Debian systems but you can open the file and see all the dependencies to install