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[cervajs at fpf.slu.cz]

hi,

is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ?

or is chan_pjsip better supported?

or the recommended way for asterisk is use respoke.io?


my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js)
chan_sip.c:10496 process_sdp: Can't provide secure audio requested in SDP offer "

sip.conf (important parts)
[vr1a882]
...
nat=force_rport,comedia
canreinvite=no
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
directmedia=yes
transport=wss,ws
dtlsrekey=60
dtlsverify=no
dtlscertfile=/etc/pki/tls/certs/rapidssl.crt
dtlsprivatekey=/etc/pki/tls/private/rapidssl.key
dtlssetup=actpass


sip dump
<--- SIP read from [url=WS:2.2.2.2:8558]WS:2.2.2.2:8558[/url] --->
INVITE [url=sip:887@ipbx]sip:887@ipbx[/url] SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKyhVsmJdAtwRvuOWz9BHiNo1DGcqp4grQ;rport
From: "cervenka"[url=sip:vr1a882@vhXXX.example.com]<sip:vr1a882@vhXXX.example.com>[/url];tag=RDmpGm2Mubc5xQQ8NMli
To: [url=sip:887@ipbx]<sip:887@ipbx>[/url]
Contact: "cervenka"[url=sips:vr1a882@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss]<sips:vr1a882@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>[/url];+g.oma.sip-im;language="en,fr"
Call-ID: cf2990ba-3f12-3d9e-adb6-52889c414ed3
CSeq: 41942 INVITE
Content-Type: application/sdp
Content-Length: 1250
Max-Forwards: 70
Authorization: Digest username="vr1a882",realm="pbx",nonce="0edd0f1f",uri=[url=sip:887@ipbx]\"sip:887@ipbx\"[/url],response="46f10b5c84accd119fea0c65bdad3dee",algorith                m=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
Organization: Doubango Telecom

v=0
o=mozilla...THIS_IS_SDPARTA-38.0.1 4294967295 0 IN IP4 127.0.0.1
s=Doubango Telecom - firefox
t=0 0
a=sendrecv
a=fingerprint:sha-256 A4:67:26:11:1F:1E:F2:8F:75:02:FE:69:2F:FC:FA:87:7A:2C:DA:86:6D:40:43:31:B7:4C:89:0B:15:44:00:56
a=group:BUNDLE sdparta_0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 52438 UDP/TLS/RTP/SAVPF 109 9 0 8
c=IN IP4 2.2.2.2
a=candidate:0 1 UDP 2128609535 10.128.3.220 52438 typ host
a=candidate:5 1 UDP 2128543999 192.168.56.1 52439 typ host
a=candidate:0 2 UDP 2128609534 10.128.3.220 52440 typ host
a=candidate:5 2 UDP 2128543998 192.168.56.1 52441 typ host
a=candidate:4 1 UDP 1692467199 2.2.2.2 52438 typ srflx raddr 10.128.3.220 rport 52438
a=candidate:4 2 UDP 1692467198 2.2.2.2 52440 typ srflx raddr 10.128.3.220 rport 52440
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=ice-pwd:5d1f4a8e35737ce88b14c471b3416e55
a=ice-ufrag:2425a6be
a=mid:sdparta_0
a=msid:{b2ea5e2d-54e2-433e-ac63-5c52874aa378} {aca6671f-8943-4773-9e36-398ef112a22f}
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=setup:actpass
a=ssrc:1181629171 cname:{dc854b06-da58-45b3-8185-bbc6a57746c0}
<------------->
--- (13 headers 31 lines) ---
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:25444 handle_request_invite: Initializing initreq for method INVITE - callid cf2990ba-                3f12-3d9e-adb6-52889c414ed3
Using INVITE request as basis request - cf2990ba-3f12-3d9e-adb6-52889c414ed3
Found peer 'vr1a882' for 'vr1a882' from 2.2.2.2:8558
[May 19 16:47:43] DEBUG[14160][C-00000007]: rtp_engine.c:421 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x9072064'
[May 19 16:47:43] DEBUG[14160][C-00000007]: res_rtp_asterisk.c:2437 ast_rtp_new: Allocated port 17304 for RTP instance '0x9072064'
[May 19 16:47:43] DEBUG[14160][C-00000007]: rtp_engine.c:430 ast_rtp_instance_new: RTP instance '0x9072064' is setup and ready to go
[May 19 16:47:43] DEBUG[14160][C-00000007]: rtp_engine.c:421 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x941ca0c'
[May 19 16:47:43] DEBUG[14160][C-00000007]: res_rtp_asterisk.c:2437 ast_rtp_new: Allocated port 12356 for RTP instance '0x941ca0c'
[May 19 16:47:43] DEBUG[14160][C-00000007]: rtp_engine.c:430 ast_rtp_instance_new: RTP instance '0x941ca0c' is setup and ready to go
[May 19 16:47:43] DEBUG[14160][C-00000007]: res_rtp_asterisk.c:4682 ast_rtp_prop_set: Setup RTCP on RTP instance '0x941ca0c'
[May 19 16:47:43] DEBUG[14160][C-00000007]: res_rtp_asterisk.c:4682 ast_rtp_prop_set: Setup RTCP on RTP instance '0x9072064'
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:5589 do_setnat: Setting NAT on RTP to On
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:5593 do_setnat: Setting NAT on VRTP to On
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10003 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10003 process_sdp: Processing session-level SDP o=mozilla...THIS_IS_SDPARTA-38.0.1 429                4967295 0 IN IP4 127.0.0.1... OK.
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10003 process_sdp: Processing session-level SDP s=Doubango Telecom - firefox... UNSUPP                ORTED OR FAILED.
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10003 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10003 process_sdp: Processing session-level SDP a=sendrecv... OK.
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:7553 initialize_udptl: Setting NAT on UDPTL to On
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10003 process_sdp: Processing session-level SDP a=fingerprint:sha-256 A4:67:26:11:1F:1                E:F2:8F:75:02:FE:69:2F:FC:FA:87:7A:2C:DA:86:6D:40:43:31:B7:4C:89:0B:15:44:00:56... UNSUPPORTED OR FAILED.
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10003 process_sdp: Processing session-level SDP a=group:BUNDLE sdparta_0... UNSUPPORTE                D OR FAILED.
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10003 process_sdp: Processing session-level SDP a=ice-options:trickle... UNSUPPORTED O                R FAILED.
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10003 process_sdp: Processing session-level SDP a=msid-semantic:WMS *... UNSUPPORTED O                R FAILED.
Found RTP audio format 109
[May 19 16:47:43] DEBUG[14160][C-00000007]: rtp_engine.c:664 ast_rtp_codecs_payloads_set_m_type: Setting payload 109 (0x951dc54) based on m t                ype on 0xb34664bc
Found RTP audio format 9
[May 19 16:47:43] DEBUG[14160][C-00000007]: rtp_engine.c:664 ast_rtp_codecs_payloads_set_m_type: Setting payload 9 (0x951efd4) based on m typ                e on 0xb34664bc
Found RTP audio format 0
[May 19 16:47:43] DEBUG[14160][C-00000007]: rtp_engine.c:664 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 (0x914f66c) based on m typ                e on 0xb34664bc
Found RTP audio format 8
[May 19 16:47:43] DEBUG[14160][C-00000007]: rtp_engine.c:664 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 (0x944dcb4) based on m typ                e on 0xb34664bc
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10458 process_sdp: Processing media-level (audio) SDP c=IN IP4 2.2.2.2... OK.
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10458 process_sdp: Processing media-level (audio) SDP a=candidate:0 1 UDP 2128609535 1                0.128.3.220 52438 typ host... OK.
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10458 process_sdp: Processing media-level (audio) SDP a=candidate:5 1 UDP 2128543999 1                92.168.56.1 52439 typ host... OK.
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10458 process_sdp: Processing media-level (audio) SDP a=candidate:0 2 UDP 2128609534 1                0.128.3.220 52440 typ host... OK.
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10458 process_sdp: Processing media-level (audio) SDP a=candidate:5 2 UDP 2128543998 1                92.168.56.1 52441 typ host... OK.
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10458 process_sdp: Processing media-level (audio) SDP a=candidate:4 1 UDP 1692467199 2                12.71.138.50 52438 typ srflx raddr 10.128.3.220 rport 52438... OK.
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10458 process_sdp: Processing media-level (audio) SDP a=candidate:4 2 UDP 1692467198 2                12.71.138.50 52440 typ srflx raddr 10.128.3.220 rport 52440... OK.
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10458 process_sdp: Processing media-level (audio) SDP a=sendrecv... OK.
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10458 process_sdp: Processing media-level (audio) SDP a=end-of-candidates... UNSUPPORT                ED OR FAILED.
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10458 process_sdp: Processing media-level (audio) SDP a=extmap:1 urn:ietf:params:rtp-h                drext:ssrc-audio-level... UNSUPPORTED OR FAILED.
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10458 process_sdp: Processing media-level (audio) SDP a=ice-pwd:5d1f4a8e35737ce88b14c4                71b3416e55... OK.
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10458 process_sdp: Processing media-level (audio) SDP a=ice-ufrag:2425a6be... OK.
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10458 process_sdp: Processing media-level (audio) SDP a=mid:sdparta_0... UNSUPPORTED O                R FAILED.
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10458 process_sdp: Processing media-level (audio) SDP a=msid:{b2ea5e2d-54e2-433e-ac63-                5c52874aa378} {aca6671f-8943-4773-9e36-398ef112a22f}... UNSUPPORTED OR FAILED.
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10458 process_sdp: Processing media-level (audio) SDP a=rtcp-mux... UNSUPPORTED OR FAI                LED.
Found audio description format opus for ID 109
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10458 process_sdp: Processing media-level (audio) SDP a=rtpmap:109 opus/48000/2... OK.
Found audio description format G722 for ID 9
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10458 process_sdp: Processing media-level (audio) SDP a=rtpmap:9 G722/8000/1... OK.
Found audio description format PCMU for ID 0
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10458 process_sdp: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
Found audio description format PCMA for ID 8
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10458 process_sdp: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10458 process_sdp: Processing media-level (audio) SDP a=setup:actpass... UNSUPPORTED O                R FAILED.
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:10458 process_sdp: Processing media-level (audio) SDP a=ssrc:1181629171 cname:{dc854b0                6-da58-45b3-8185-bbc6a57746c0}... UNSUPPORTED OR FAILED.
[May 19 16:47:43] WARNING[14160][C-00000007]: chan_sip.c:10496 process_sdp: Can't provide secure audio requested in SDP offer

<--- Reliably Transmitting (NAT) to 2.2.2.2:8558 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKyhVsmJdAtwRvuOWz9BHiNo1DGcqp4grQ;received=2.2.2.2;rport=8558
From: "cervenka"[url=sip:vr1a882@vhXXX.example.com]<sip:vr1a882@vhXXX.example.com>[/url];tag=RDmpGm2Mubc5xQQ8NMli
To: [url=sip:887@ipbx]<sip:887@ipbx>[/url];tag=as5d30f0ef
Call-ID: cf2990ba-3f12-3d9e-adb6-52889c414ed3
CSeq: 41942 INVITE
Server: ipbx 3.2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:3696 __sip_xmit: Trying to put 'SIP/2.0 488' onto WS socket destined for 2.2.2.2                :8558
Scheduling destruction of SIP dialog 'cf2990ba-3f12-3d9e-adb6-52889c414ed3' in 32000 ms (Method: INVITE)
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:25557 handle_request_invite: No compatible codecs for this SIP call.
[May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:28297 handle_request_do: SIP message could not be handled, bad request: cf2990ba-3f12-                3d9e-adb6-52889c414ed3

[62mkv at mail.ru]

Hi Marek

Yes, here is a person with (mostly) working Asterisk 13 (chan_sip) +
WebRTC (using sipml5 js lib) setup

You can contact me directly, if you wish, I will try to help if I can

As of the issue you have.. is it because you're working with FF 37 as
browser ?

I have not come across such issues since last summer, when FF (or
Asterisk, don't remember exactly) had problems with proper DTLS-SRTP
implementation

Yours, Kirill

22.05.2015 23:00, asterisk-users-request@lists.digium.com пишет: