Opus codec deploy in asterik made unable to call establish

Asterisk Asterisk Support
#1

Hello Experts,

I was running a asterisk 11.5 with no error. After installed Opus codec with patch in my asterisk
the latest codecs are:
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
ID TYPE NAME DESCRIPTION

100001 audio g723 (G.723.1)
100002 audio gsm (GSM)
100003 audio ulaw (G.711 u-law)
100004 audio alaw (G.711 A-law)
100011 audio g726 (G.726 RFC3551)
100006 audio adpcm (ADPCM)
100019 audio slin (16 bit Signed Linear PCM)
100007 audio lpc10 (LPC10)
100008 audio g729 (G.729A)
100009 audio speex (SpeeX)
100016 audio speex16 (SpeeX 16khz)
100010 audio ilbc (iLBC)
100005 audio g726aal2 (G.726 AAL2)
100012 audio g722 (G722)
100021 audio slin16 (16 bit Signed Linear PCM (16kHz))
300001 image jpeg (JPEG image)
300002 image png (PNG image)
200001 video h261 (H.261 Video)
200002 video h263 (H.263 Video)
200003 video h263p (H.263+ Video)
200004 video h264 (H.264 Video)
200005 video mpeg4 (MPEG4 Video)
400001 text red (T.140 Realtime Text with redundancy)
400002 text t140 (Passthrough T.140 Realtime Text)
100013 audio siren7 (ITU G.722.1 (Siren7, licensed from Polycom))
100014 audio siren14 (ITU G.722.1 Annex C, (Siren14, licensed from Polycom))
100017 audio testlaw (G.711 test-law)
100015 audio g719 (ITU G.719)
100028 audio speex32 (SpeeX 32khz)
100020 audio slin12 (16 bit Signed Linear PCM (12kHz))
100022 audio slin24 (16 bit Signed Linear PCM (24kHz))
100023 audio slin32 (16 bit Signed Linear PCM (32kHz))
100024 audio slin44 (16 bit Signed Linear PCM (44kHz))
100025 audio slin48 (16 bit Signed Linear PCM (48kHz))
100026 audio slin96 (16 bit Signed Linear PCM (96kHz))
100027 audio slin192 (16 bit Signed Linear PCM (192kHz))
100030 audio opus (Opus Codec)
200006 video vp8 (VP8 Video)
100018 audio silk8 (SILK Custom Format 8khz)
100018 audio silk12 (SILK Custom Format 12khz)
100018 audio silk16 (SILK Custom Format 16khz)
100018 audio silk24 (SILK Custom Format 24khz)

Now whenever I am making audio call between 2 peer, no call is getting establish and throwing below errors:

[btw i am using browser based sip softphone jssip]

[Sep 29 09:27:36] WARNING[17065][C-00000003]: sip/sdp_crypto.c:173 sdp_crypto_activate: Could not set SRTP policies
[Sep 29 09:27:36] WARNING[17065][C-00000003]: sip/sdp_crypto.c:173 sdp_crypto_activate: Could not set SRTP policies
[Sep 29 09:27:36] WARNING[17065][C-00000003]: chan_sip.c:11100 process_sdp_a_audio: Got Opus minptime=10
[Sep 29 09:27:36] WARNING[17065][C-00000003]: chan_sip.c:10437 process_sdp: Rejecting secure audio stream without encryption details: audio 17002 RTP/SAVPF 111 103 104 0 8 106 105 13 126

[1000]
secret=test123
context=local
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
qualify=yes
qualifyfreq=600
transport=udp,wss,ws
encryption=yes
dial=SIP/8000
callerid=testuser <8000>
callcounter=yes
avpf=yes
icesupport=yes
directmedia=no

Can any one cooperate?

#2

Seems like your webrtc client isn’t setting the secure layer. The last warning said that.

  • On the asterisk side check if libsrtp is loaded
  • Check your webrtc settings
  • About opus I don’t know if it has a conf file, if so check if you can change ptime prime 20.
#3

Thanks!

  1. srtp is enabled. Actually asterisk configure with srtp.
    Asterisk is compiled as

./configure --with-crypto --with-ssl=ssl --with-srtp

  1. The webrtc client is jssip. I asked them and they replied
    "Not a JsSIP issue. Please ask in Asterisk forums." Here is the thread:(though that asked for video call, fact is same for video and audio)
    groups.google.com/forum/#!topic … XKLzDfVId0
#4

That’s why I prefer sipml5 over jssip. That guy always respond with that. You need to enable the sip debug to see the sdp transaction. But asterisk is complaining about the rtp isn’t secure. So lets check what side has the error.

#5

Hey

I have found something. What i did:

i have changed the peer setting on transport=wss as below:

[1000]
secret=test123
context=local
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
qualify=yes
qualifyfreq=600
transport=wss
encryption=yes
dial=SIP/8000
callerid=testuser <8000>
callcounter=yes
avpf=yes
icesupport=yes
directmedia=no

made call and got below message from asterisk CLI:

[Sep 29 12:02:45] WARNING[18352][C-00000007]: sip/sdp_crypto.c:173 sdp_crypto_activate: Could not set SRTP policies
[Sep 29 12:02:45] WARNING[18352][C-00000007]: sip/sdp_crypto.c:173 sdp_crypto_activate: Could not set SRTP policies
[Sep 29 12:02:45] WARNING[18352][C-00000007]: chan_sip.c:11100 process_sdp_a_audio: Got Opus minptime=10
[Sep 29 12:02:45] WARNING[18352][C-00000007]: chan_sip.c:10437 process_sdp: Rejecting secure audio stream without encryption details: audio 17101 RTP/SAVPF 111 103 104 0 8 106 105 13 126

Should i need to relaod the srtp again? If yes how i can do that?

#6

Enable the sip debug to see why isn’t setting that. To reload the module use module reload modulename.so( I dont have the exact name right now).

#7

okay i will enable the log and share. by the mean time i tried:

smplsrv*CLI> module reload res_srtp.so
Module ‘res_srtp.so’ does not support reload

:frowning:

#8

Then try module unload res_srtp.so after that module load res_srtp.so

#9

Hi

Please find the SIP debug using below sip Extension setting:

[1000]
secret=test123
context=local
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
qualify=yes
qualifyfreq=600
transport=udp,ws
encryption=yes
dial=SIP/8000
callerid=testuser <8000>
callcounter=yes
avpf=yes
icesupport=yes
directmedia=no

================[Debug CLI]=========================

Via: SIP/2.0/WS 9b307lcmnb89.invalid;branch=z9hG4bK2939020
Max-Forwards: 69
To: sip:9000@65.51.243.124
From: “Mahfuz” sip:8000@65.51.243.124;tag=iqr5rj2bsq
Call-ID: i2g4ju4076ce0fe9pq4p
CSeq: 257 INVITE
Contact: sip:me39fbhc@9b307lcmnb89.invalid;transport=ws;ob
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
Content-Type: application/sdp
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 3756

v=0
o=- 6189261762902729741 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS zPOTsGAQ5XHJ4zxHrxJqW1ZwUEofB02hWLOp
m=audio 17039 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 180.234.107.73
a=rtcp:17039 IN IP4 180.234.107.73
a=candidate:2045219518 1 udp 2113937151 10.37.131.2 58647 typ host generation 0
a=candidate:2045219518 2 udp 2113937151 10.37.131.2 58647 typ host generation 0
a=candidate:2026329132 1 udp 2113937151 192.168.0.129 58648 typ host generation 0
a=candidate:2026329132 2 udp 2113937151 192.168.0.129 58648 typ host generation 0
a=candidate:678703848 1 udp 2113937151 192.168.100.1 58649 typ host generation 0
a=candidate:678703848 2 udp 2113937151 192.168.100.1 58649 typ host generation 0
a=candidate:3988789767 1 udp 2113937151 192.168.100.9 58650 typ host generation 0
a=candidate:3988789767 2 udp 2113937151 192.168.100.9 58650 typ host generation 0
a=candidate:3350409123 1 udp 2113937151 192.168.0.101 58651 typ host generation 0
a=candidate:3350409123 2 udp 2113937151 192.168.0.101 58651 typ host generation 0
a=candidate:2998616347 1 udp 2113937151 10.37.130.2 58652 typ host generation 0
a=candidate:2998616347 2 udp 2113937151 10.37.130.2 58652 typ host generation 0
a=candidate:2803279295 1 udp 2113937151 192.168.100.5 58653 typ host generation 0
a=candidate:2803279295 2 udp 2113937151 192.168.100.5 58653 typ host generation 0
a=candidate:929328718 1 tcp 1509957375 10.37.131.2 0 typ host generation 0
a=candidate:929328718 2 tcp 1509957375 10.37.131.2 0 typ host generation 0
a=candidate:910469340 1 tcp 1509957375 192.168.0.129 0 typ host generation 0
a=candidate:910469340 2 tcp 1509957375 192.168.0.129 0 typ host generation 0
a=candidate:1727516184 1 tcp 1509957375 192.168.100.1 0 typ host generation 0
a=candidate:1727516184 2 tcp 1509957375 192.168.100.1 0 typ host generation 0
a=candidate:2739136247 1 tcp 1509957375 192.168.100.9 0 typ host generation 0
a=candidate:2739136247 2 tcp 1509957375 192.168.100.9 0 typ host generation 0
a=candidate:2301678419 1 tcp 1509957375 192.168.0.101 0 typ host generation 0
a=candidate:2301678419 2 tcp 1509957375 192.168.0.101 0 typ host generation 0
a=candidate:4231577067 1 tcp 1509957375 10.37.130.2 0 typ host generation 0
a=candidate:4231577067 2 tcp 1509957375 10.37.130.2 0 typ host generation 0
a=candidate:3918879055 1 tcp 1509957375 192.168.100.5 0 typ host generation 0
a=candidate:3918879055 2 tcp 1509957375 192.168.100.5 0 typ host generation 0
a=candidate:1190865175 1 udp 1845501695 180.234.107.73 17039 typ srflx raddr 192.168.0.101 rport 58651 generation 0
a=candidate:1190865175 2 udp 1845501695 180.234.107.73 17039 typ srflx raddr 192.168.0.101 rport 58651 generation 0
a=ice-ufrag:299rNEBo/vH30Oup
a=ice-pwd:TawByAxY5s6RAVf/VCqrAJSs
a=ice-options:google-ice
a=fingerprint:sha-256 CD:A6:92:45:D1:D3:21:4B:47:48:67:32:FC:86:21:0C:CD:30:52:1E:AC:C1:97:CA:5A:39:60:22:04:3E:CC:E6
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:/ZDQWuDa9Jf6FWjG2kz3DpU35hs95CKnQ02Iy9p1
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:XVfgzmX8taUwUFqqAeYIcmeJBLVYiYRsXN/MBPv/
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:870364229 cname:wvjQ1xeli9duYld1
a=ssrc:870364229 msid:zPOTsGAQ5XHJ4zxHrxJqW1ZwUEofB02hWLOp zPOTsGAQ5XHJ4zxHrxJqW1ZwUEofB02hWLOpa0
a=ssrc:870364229 mslabel:zPOTsGAQ5XHJ4zxHrxJqW1ZwUEofB02hWLOp
a=ssrc:870364229 label:zPOTsGAQ5XHJ4zxHrxJqW1ZwUEofB02hWLOpa0
<------------->
— (13 headers 65 lines) —
Using INVITE request as basis request - i2g4ju4076ce0fe9pq4p
Found peer ‘8000’ for ‘8000’ from 180.234.107.73:17035

<— Reliably Transmitting (no NAT) to 180.234.107.73:5060 —>
SIP/2.0 401 Unauthorized
v: SIP/2.0/WS 9b307lcmnb89.invalid;branch=z9hG4bK2939020;received=180.234.107.73
f: “Mahfuz” sip:8000@65.51.243.124;tag=iqr5rj2bsq
t: sip:9000@65.51.243.124;tag=as19a58742
i: i2g4ju4076ce0fe9pq4p
CSeq: 257 INVITE
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
k: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="377157e9"
l: 0

<------------>
Scheduling destruction of SIP dialog ‘i2g4ju4076ce0fe9pq4p’ in 89472 ms (Method: INVITE)

<— SIP read from WS:180.234.107.73:17035 —>
ACK sip:9000@65.51.243.124 SIP/2.0
Via: SIP/2.0/WS 9b307lcmnb89.invalid;branch=z9hG4bK2939020
To: sip:9000@65.51.243.124;tag=as19a58742
From: “Mahfuz” sip:8000@65.51.243.124;tag=iqr5rj2bsq
Call-ID: i2g4ju4076ce0fe9pq4p
CSeq: 257 ACK

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

<— SIP read from WS:180.234.107.73:17035 —>
INVITE sip:9000@65.51.243.124 SIP/2.0
Via: SIP/2.0/WS 9b307lcmnb89.invalid;branch=z9hG4bK5414977
Max-Forwards: 69
To: sip:9000@65.51.243.124
From: “Mahfuz” sip:8000@65.51.243.124;tag=iqr5rj2bsq
Call-ID: i2g4ju4076ce0fe9pq4p
CSeq: 258 INVITE
Authorization: Digest algorithm=MD5, username=“8000”, realm=“asterisk”, nonce=“377157e9”, uri="sip:9000@65.51.243.124", response="fc3399f439f9917c6da69d577783d409"
Contact: sip:me39fbhc@9b307lcmnb89.invalid;transport=ws;ob
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
Content-Type: application/sdp
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 3756

v=0
o=- 6189261762902729741 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS zPOTsGAQ5XHJ4zxHrxJqW1ZwUEofB02hWLOp
m=audio 17039 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 180.234.107.73
a=rtcp:17039 IN IP4 180.234.107.73
a=candidate:2045219518 1 udp 2113937151 10.37.131.2 58647 typ host generation 0
a=candidate:2045219518 2 udp 2113937151 10.37.131.2 58647 typ host generation 0
a=candidate:2026329132 1 udp 2113937151 192.168.0.129 58648 typ host generation 0
a=candidate:2026329132 2 udp 2113937151 192.168.0.129 58648 typ host generation 0
a=candidate:678703848 1 udp 2113937151 192.168.100.1 58649 typ host generation 0
a=candidate:678703848 2 udp 2113937151 192.168.100.1 58649 typ host generation 0
a=candidate:3988789767 1 udp 2113937151 192.168.100.9 58650 typ host generation 0
a=candidate:3988789767 2 udp 2113937151 192.168.100.9 58650 typ host generation 0
a=candidate:3350409123 1 udp 2113937151 192.168.0.101 58651 typ host generation 0
a=candidate:3350409123 2 udp 2113937151 192.168.0.101 58651 typ host generation 0
a=candidate:2998616347 1 udp 2113937151 10.37.130.2 58652 typ host generation 0
a=candidate:2998616347 2 udp 2113937151 10.37.130.2 58652 typ host generation 0
a=candidate:2803279295 1 udp 2113937151 192.168.100.5 58653 typ host generation 0
a=candidate:2803279295 2 udp 2113937151 192.168.100.5 58653 typ host generation 0
a=candidate:929328718 1 tcp 1509957375 10.37.131.2 0 typ host generation 0
a=candidate:929328718 2 tcp 1509957375 10.37.131.2 0 typ host generation 0
a=candidate:910469340 1 tcp 1509957375 192.168.0.129 0 typ host generation 0
a=candidate:910469340 2 tcp 1509957375 192.168.0.129 0 typ host generation 0
a=candidate:1727516184 1 tcp 1509957375 192.168.100.1 0 typ host generation 0
a=candidate:1727516184 2 tcp 1509957375 192.168.100.1 0 typ host generation 0
a=candidate:2739136247 1 tcp 1509957375 192.168.100.9 0 typ host generation 0
a=candidate:2739136247 2 tcp 1509957375 192.168.100.9 0 typ host generation 0
a=candidate:2301678419 1 tcp 1509957375 192.168.0.101 0 typ host generation 0
a=candidate:2301678419 2 tcp 1509957375 192.168.0.101 0 typ host generation 0
a=candidate:4231577067 1 tcp 1509957375 10.37.130.2 0 typ host generation 0
a=candidate:4231577067 2 tcp 1509957375 10.37.130.2 0 typ host generation 0
a=candidate:3918879055 1 tcp 1509957375 192.168.100.5 0 typ host generation 0
a=candidate:3918879055 2 tcp 1509957375 192.168.100.5 0 typ host generation 0
a=candidate:1190865175 1 udp 1845501695 180.234.107.73 17039 typ srflx raddr 192.168.0.101 rport 58651 generation 0
a=candidate:1190865175 2 udp 1845501695 180.234.107.73 17039 typ srflx raddr 192.168.0.101 rport 58651 generation 0
a=ice-ufrag:299rNEBo/vH30Oup
a=ice-pwd:TawByAxY5s6RAVf/VCqrAJSs
a=ice-options:google-ice
a=fingerprint:sha-256 CD:A6:92:45:D1:D3:21:4B:47:48:67:32:FC:86:21:0C:CD:30:52:1E:AC:C1:97:CA:5A:39:60:22:04:3E:CC:E6
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:/ZDQWuDa9Jf6FWjG2kz3DpU35hs95CKnQ02Iy9p1
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:XVfgzmX8taUwUFqqAeYIcmeJBLVYiYRsXN/MBPv/
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:870364229 cname:wvjQ1xeli9duYld1
a=ssrc:870364229 msid:zPOTsGAQ5XHJ4zxHrxJqW1ZwUEofB02hWLOp zPOTsGAQ5XHJ4zxHrxJqW1ZwUEofB02hWLOpa0
a=ssrc:870364229 mslabel:zPOTsGAQ5XHJ4zxHrxJqW1ZwUEofB02hWLOp
a=ssrc:870364229 label:zPOTsGAQ5XHJ4zxHrxJqW1ZwUEofB02hWLOpa0
<------------->
— (14 headers 65 lines) —
Using INVITE request as basis request - i2g4ju4076ce0fe9pq4p
Found peer ‘8000’ for ‘8000’ from 180.234.107.73:17035
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 126
[Sep 30 03:36:41] WARNING[19688][C-00000075]: sip/sdp_crypto.c:173 sdp_crypto_activate: Could not set SRTP policies
[Sep 30 03:36:41] WARNING[19688][C-00000075]: sip/sdp_crypto.c:173 sdp_crypto_activate: Could not set SRTP policies
Found audio description format opus for ID 111
[Sep 30 03:36:41] WARNING[19688][C-00000075]: chan_sip.c:11100 process_sdp_a_audio: Got Opus minptime=10
Found unknown media description format ISAC for ID 103
Found unknown media description format ISAC for ID 104
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 106
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found audio description format telephone-event for ID 126
[Sep 30 03:36:41] WARNING[19688][C-00000075]: chan_sip.c:10437 process_sdp: Rejecting secure audio stream without encryption details: audio 17039 RTP/SAVPF 111 103 104 0 8 106 105 13 126

<— Reliably Transmitting (no NAT) to 180.234.107.73:5060 —>
SIP/2.0 488 Not acceptable here
v: SIP/2.0/WS 9b307lcmnb89.invalid;branch=z9hG4bK5414977;received=180.234.107.73
f: “Mahfuz” sip:8000@65.51.243.124;tag=iqr5rj2bsq
t: sip:9000@65.51.243.124;tag=as19a58742
i: i2g4ju4076ce0fe9pq4p
CSeq: 258 INVITE
Server: Asterisk PBX 11.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
k: replaces, timer
l: 0

<------------>
Scheduling destruction of SIP dialog ‘i2g4ju4076ce0fe9pq4p’ in 89472 ms (Method: INVITE)

<— SIP read from WS:180.234.107.73:17035 —>
ACK sip:9000@65.51.243.124 SIP/2.0
Via: SIP/2.0/WS 9b307lcmnb89.invalid;branch=z9hG4bK5414977
To: sip:9000@65.51.243.124;tag=as19a58742
From: “Mahfuz” sip:8000@65.51.243.124;tag=iqr5rj2bsq
Call-ID: i2g4ju4076ce0fe9pq4p
CSeq: 258 ACK

<------------->
— (6 headers 0 lines) —
Really destroying SIP dialog ‘csqfc2a3fno3mqeu02qvv0’ Method: REGISTER
Really destroying SIP dialog ‘7d26f9k82cgu7lohqa381a’ Method: REGISTER
Really destroying SIP dialog ‘df1f76kdog1151te86sb’ Method: INVITE
smplsrv*CLI>

#10

Hi

My Chrome browser is latest and updated. tried with :

chrome: stable Version 29.0.1547.76 m
chrome: Version 32.0.1655.0 canary Aura

I have clean all. Install SRTP and asterisk freshly then also tried siplm5(webrtc2sip) web phone latest release.

found the same error:

root@smplsrv:/# asterisk -rvvvvvvvvvvvvvvvvv
Asterisk 11.4.0, Copyright © 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

Connected to Asterisk SVN-trunk-r379070M currently running on smplsrv (pid = 18437)
== WebSocket connection from ‘180.234.107.73:17009’ for protocol ‘sip’ accepted using version ‘13’
== WebSocket connection from ‘180.234.107.73:17006’ closed
– Registered SIP ‘8000’ at 180.234.107.73:17009
> Saved useragent “IM-client/OMA1.0 sipML5-v1.2013.08.10B” for peer 8000
== WebSocket connection from ‘180.234.107.73:17008’ for protocol ‘sip’ accepted using version ‘13’
== WebSocket connection from ‘180.234.107.73:17005’ closed
– Registered SIP ‘9000’ at 180.234.107.73:17008
> Saved useragent “IM-client/OMA1.0 sipML5-v1.2013.08.10B” for peer 9000
== Using SIP RTP CoS mark 5
[Sep 30 04:33:09] WARNING[18706][C-00000006]: sip/sdp_crypto.c:172 sdp_crypto_activate: Could not set SRTP policies
[Sep 30 04:33:09] WARNING[18706][C-00000006]: sip/sdp_crypto.c:172 sdp_crypto_activate: Could not set SRTP policies
[Sep 30 04:33:09] WARNING[18706][C-00000006]: chan_sip.c:11155 process_sdp_a_audio: Got Opus minptime=10
[Sep 30 04:33:09] WARNING[18706][C-00000006]: chan_sip.c:10492 process_sdp: Rejecting secure audio stream without encryption details: audio 17018 RTP/SAVPF 111 103 104 0 8 106 105 13 126
smplsrv*CLI>

What i need to do change in asterisk or just guessing chrome browser is not sending secured SDP or what?

#11

Also just checked with latest firefox latest browser. Result same:

<— SIP read from WS:180.234.107.73:17028 —>
INVITE sip:9000@65.51.243.124 SIP/2.0
Via: SIP/2.0/WS jq7e0d9t23r2.invalid;branch=z9hG4bK898697
Max-Forwards: 69
To: sip:9000@65.51.243.124
From: “Mahfuz” sip:8000@65.51.243.124;tag=r5rltjfiup
Call-ID: bfrhpdisbbbabvlvodqh
CSeq: 550 INVITE
Authorization: Digest algorithm=MD5, username=“8000”, realm=“asterisk”, nonce=“32803e32”, uri="sip:9000@65.51.243.124", response="6e9565c9cdde3971933214c7689db583"
Contact: sip:pd5lgfl1@jq7e0d9t23r2.invalid;transport=ws;ob
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE
Content-Type: application/sdp
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 1314

v=0
o=Mozilla-SIPUA-23.0.1 24143 0 IN IP4 0.0.0.0
s=SIP Call
t=0 0
a=ice-ufrag:84e61d02
a=ice-pwd:95fade6f8102b01352c8d063e1687072
a=fingerprint:sha-256 A2:81:DD:04:DB:80:2C:97:8F:AE:18:2A:DA:07:13:D3:F7:B5:CF:41:86:C3:A3:CC:44:4A:5D:72:4D:AB:43:2B
m=audio 61117 RTP/SAVPF 109 0 8 101
c=IN IP4 192.168.0.101
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:0 1 UDP 2111832319 192.168.0.101 61117 typ host
a=candidate:2 1 UDP 2111766783 10.37.130.2 61118 typ host
a=candidate:4 1 UDP 2111766783 192.168.100.5 61119 typ host
a=candidate:6 1 UDP 2111701247 10.37.131.2 61120 typ host
a=candidate:8 1 UDP 2111701247 192.168.0.129 61121 typ host
a=candidate:10 1 UDP 2111701247 192.168.100.1 61122 typ host
a=candidate:12 1 UDP 2111701247 192.168.100.9 61123 typ host
a=candidate:0 2 UDP 2111832318 192.168.0.101 61124 typ host
a=candidate:2 2 UDP 2111766782 10.37.130.2 61125 typ host
a=candidate:4 2 UDP 2111766782 192.168.100.5 61126 typ host
a=candidate:6 2 UDP 2111701246 10.37.131.2 61127 typ host
a=candidate:8 2 UDP 2111701246 192.168.0.129 61128 typ host
a=candidate:10 2 UDP 2111701246 192.168.100.1 61129 typ host
a=candidate:12 2 UDP 2111701246 192.168.100.9 61130 typ host
<------------->
— (14 headers 30 lines) —
Using INVITE request as basis request - bfrhpdisbbbabvlvodqh
Found peer ‘8000’ for ‘8000’ from 180.234.107.73:17028
== Using SIP RTP CoS mark 5
Found RTP audio format 109
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format opus for ID 109
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
[Sep 30 04:59:25] WARNING[18725][C-00000009]: chan_sip.c:10492 process_sdp: Rejecting secure audio stream without encryption details: audio 61117 RTP/SAVPF 109 0 8 101

<— Reliably Transmitting (no NAT) to 180.234.107.73:5060 —>
SIP/2.0 488 Not acceptable here
v: SIP/2.0/WS jq7e0d9t23r2.invalid;branch=z9hG4bK898697;received=180.234.107.73
f: “Mahfuz” sip:8000@65.51.243.124;tag=r5rltjfiup
t: sip:9000@65.51.243.124;tag=as54fc6255
i: bfrhpdisbbbabvlvodqh
CSeq: 550 INVITE
Server: Asterisk PBX SVN-trunk-r379070M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
k: replaces, timer
l: 0

#12

Log from chrome console:

To: sip:8000@65.51.243.124
Contact: "ira"sip:9000@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws;impi=9000;ha1=c02298b31af0729048774a673c440ff6;+g.oma.sip-im;+sip.ice;language=“en,fr"
Call-ID: 6874a915-2a24-9ef7-d1c7-b9c8213f85e9
CSeq: 17821 INVITE
Content-Type: application/sdp
Content-Length: 3784
Route: sip:65.51.243.124:5060;lr;sipml5-outbound;transport=udp
Max-Forwards: 70
Authorization: Digest username=“9000”,realm=“asterisk”,nonce=“518ca9e5”,uri="sip:8000@65.51.243.124”,response=“5699801dd7eb8fd84dd78b83b99425e7”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.08.10B
Organization: Doubango Telecom

v=0
o=- 8536183350611728000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS FGEl8a9W35qlKNcVD9cD3806mCssreDDKxYQ
m=audio 17010 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 180.234.107.73
a=rtcp:17010 IN IP4 180.234.107.73
a=candidate:2045219518 1 udp 2113937151 10.37.131.2 56438 typ host generation 0
a=candidate:2045219518 2 udp 2113937151 10.37.131.2 56438 typ host generation 0
a=candidate:2026329132 1 udp 2113937151 192.168.0.129 56439 typ host generation 0
a=candidate:2026329132 2 udp 2113937151 192.168.0.129 56439 typ host generation 0
a=candidate:678703848 1 udp 2113937151 192.168.100.1 56440 typ host generation 0
a=candidate:678703848 2 udp 2113937151 192.168.100.1 56440 typ host generation 0
a=candidate:3988789767 1 udp 2113937151 192.168.100.9 56441 typ host generation 0
a=candidate:3988789767 2 udp 2113937151 192.168.100.9 56441 typ host generation 0
a=candidate:3350409123 1 udp 2113937151 192.168.0.101 56442 typ host generation 0
a=candidate:3350409123 2 udp 2113937151 192.168.0.101 56442 typ host generation 0
a=candidate:2998616347 1 udp 2113937151 10.37.130.2 56443 typ host generation 0
a=candidate:2998616347 2 udp 2113937151 10.37.130.2 56443 typ host generation 0
a=candidate:2803279295 1 udp 2113937151 192.168.100.5 56444 typ host generation 0
a=candidate:2803279295 2 udp 2113937151 192.168.100.5 56444 typ host generation 0
a=candidate:929328718 1 tcp 1509957375 10.37.131.2 0 typ host generation 0
a=candidate:929328718 2 tcp 1509957375 10.37.131.2 0 typ host generation 0
a=candidate:910469340 1 tcp 1509957375 192.168.0.129 0 typ host generation 0
a=candidate:910469340 2 tcp 1509957375 192.168.0.129 0 typ host generation 0
a=candidate:1727516184 1 tcp 1509957375 192.168.100.1 0 typ host generation 0
a=candidate:1727516184 2 tcp 1509957375 192.168.100.1 0 typ host generation 0
a=candidate:2739136247 1 tcp 1509957375 192.168.100.9 0 typ host generation 0
a=candidate:2739136247 2 tcp 1509957375 192.168.100.9 0 typ host generation 0
a=candidate:2301678419 1 tcp 1509957375 192.168.0.101 0 typ host generation 0
a=candidate:2301678419 2 tcp 1509957375 192.168.0.101 0 typ host generation 0
a=candidate:4231577067 1 tcp 1509957375 10.37.130.2 0 typ host generation 0
a=candidate:4231577067 2 tcp 1509957375 10.37.130.2 0 typ host generation 0
a=candidate:3918879055 1 tcp 1509957375 192.168.100.5 0 typ host generation 0
a=candidate:3918879055 2 tcp 1509957375 192.168.100.5 0 typ host generation 0
a=candidate:1190865175 1 udp 1845501695 180.234.107.73 17010 typ srflx raddr 192.168.0.101 rport 56442 generation 0
a=candidate:1190865175 2 udp 1845501695 180.234.107.73 17010 typ srflx raddr 192.168.0.101 rport 56442 generation 0
a=ice-ufrag:+G3P+aDaQ0onjHvl
a=ice-pwd:L7JAHtfvoYP1ulJCTRL8WxDg
a=ice-options:google-ice
a=fingerprint:sha-256 02:1D:D8:49:CB:FD:7A:49:AE:7B:0D:27:7B:63:E8:B9:1B:52:6C:8C:BE:08:AD:D6:7D:8F:60:0D:51:33:BF:99
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:575MtoHpho94NsX+Yc4BRCNz89cLo9BOtIFJZneX
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:d8a/FCx+C6elTuMpxvEQTpk/6zvi3vZ4mhi6jp63
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:2823102743 cname:Wh1d2maMRkGYWmLM
a=ssrc:2823102743 msid:FGEl8a9W35qlKNcVD9cD3806mCssreDDKxYQ FGEl8a9W35qlKNcVD9cD3806mCssreDDKxYQa0
a=ssrc:2823102743 mslabel:FGEl8a9W35qlKNcVD9cD3806mCssreDDKxYQ
a=ssrc:2823102743 label:FGEl8a9W35qlKNcVD9cD3806mCssreDDKxYQa0
__tsip_transport_ws_onmessage
recv=SIP/2.0 488 Not acceptable here
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=180.234.107.73;branch=z9hG4bK83GECmMt2eHXptL6GTbdpvxJSQFIt02h
From: "ira"sip:9000@65.51.243.124;tag=08ya7T7orkQqQwqa6ZrV
To: sip:8000@65.51.243.124;tag=as342813c5
Call-ID: 6874a915-2a24-9ef7-d1c7-b9c8213f85e9
CSeq: 17821 INVITE
Content-Length: 0
Server: Asterisk PBX SVN-trunk-r379070M
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer

SEND: ACK sip:8000@65.51.243.124 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK83GECmMt2eHXptL6GTbdpvxJSQFIt02h;rport
From: "ira"sip:9000@65.51.243.124;tag=08ya7T7orkQqQwqa6ZrV
To: sip:8000@65.51.243.124;tag=as342813c5
Call-ID: 6874a915-2a24-9ef7-d1c7-b9c8213f85e9
CSeq: 17821 ACK
Content-Length: 0
Route: sip:65.51.243.124:5060;lr;sipml5-outbound;transport=udp
Max-Forwards: 70

State machine: c0000_Outgoing_2_Terminated_X_i300_to_i699INVITE
=== INVITE Dialog terminated ===
PeerConnection::stop()
==session event = i_ao_request
==session event = terminated
The FSM is in the final state
State machine: tsip_dialog_register_Connected_2_InProgress_X_oRegister
SEND: REGISTER sip:65.51.243.124 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKSCDkqHqHkFBfRyWc3mMi6urfn23axyKi;rport
From: "ira"sip:9000@65.51.243.124;tag=IBQZT83INTefehXDtFR9
To: "ira"sip:9000@65.51.243.124
Contact: "ira"sip:9000@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 1250bf28-adfe-2888-12ae-5048afc7a52f
CSeq: 60863 REGISTER
Content-Length: 0

#13

It is objecting to RTP/SAVPF, without any session key information, not to the codec.

#14

What i need to do the change in rtp/savp?

right now srtp is loaded with asterisk. I unload the srtp module and tried result same. Any guide?

#15

You need to change the client to either not request SRTP, or to provide the session key when it makes the request.

#16

I have tried both:

https://code.google.com/p/sipml5/wiki/Asterisk
http://tryit.jssip.net/

same result. both are working on websocket(ws)

#17

You need to setup both clients to use srtp try configuring your clients with wss or disable the encryption on asterisk side. As I told you in my first post the issue is your webrtc client and not asterisk.

#18

Thanks! for the reply.

First i want to try with disable the encryption on asterisk. What i need to do?

  1. Remove SRTP
  2. in sip.conf
    encryption = false
    avpf=yes

or what?

#19

comment or remove the line: encryption=yes

#20

I have comment out
;encryption=yes

tried call and the result same:

[Sep 30 19:34:17] WARNING[16327][C-0000000c]: sip/sdp_crypto.c:172 sdp_crypto_activate: Could not set SRTP policies
[Sep 30 19:34:17] WARNING[16327][C-0000000c]: sip/sdp_crypto.c:172 sdp_crypto_activate: Could not set SRTP policies
[Sep 30 19:34:17] WARNING[16327][C-0000000c]: chan_sip.c:11155 process_sdp_a_audio: Got Opus minptime=10
[Sep 30 19:34:17] WARNING[16327][C-0000000c]: chan_sip.c:10492 process_sdp: Rejecting secure audio stream without encryption details: audio 17079 RTP/SAVPF 111 103 104 0 8 106 105 13 126

Should i also need remove srtp?
Or clean srtp from system and install asterisk from scratch?