Asterisk realtime, webrtc2sip & SipML5

charlesc · May 12, 2015, 2:28pm

Hi,

I’ve asked about this on the webrtc2sip forum, but it looks like it relates to an error in my asterisk setup.

The setup is as follows:

Browser (firefox) -> webrtc2sip ->asterisk (13.3.2) ->SFLPhone on Ubuntu

SFLPhone is set up with SRTP key exchange via SDES, and ‘fallback on RTP on SDES failure’.

If I place a call from the FF browser to the SFLPhone the call is placed, but then dropped when the call is answered at the remote end. A warning appears on the asterisk console:

[May 6 16:46:44] WARNING[20191][C-0000000d]: chan_sip.c:10484 process_sdp: Failing due to no acceptable offer found

I’ve got two problems here

I haven’t been able to place a call to any other softphone (Linphone, Ekiga) at all. Only SFLPhone gets as far as setting up the call, and that seems to be because it supports SDES
I can only place a call from SFLPhone to the browser, not from the browser back to SFLPhone.

I think what’s happening is the configuration wants the connection to be encrypted from end to end - from the softphone to the browser. I need to be able to support SIP softphones that don’t have encryption though. I understand that the webrtc standards require the browser->webrtc2sip connection to be encrypted, but can’t the connection from Asterisk to the softphone be left unencrypted?

Could someone suggest where I’m going wrong?

thanks,

Charles

output from console with sip debug on


<--- SIP read from UDP:192.168.1.241:10060 --->
INVITE sip:60007@192.168.1.241 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.241:10060;branch=z9hG4bK-646837569;rport
From: <sip:60002@192.168.1.241>;tag=2080760246
To: <sip:60007@192.168.1.241>
Contact: <sip:60002@192.168.1.241:10060;ws-src-ip=192.168.1.4;ws-src-port=60457;ws-src-proto=ws;transport=udp>
Call-ID: a66c6532-4453-2020-9839-3c44e616a9db
CSeq: 2017927504 INVITE
Content-Type: application/sdp
Content-Length: 1753
Max-Forwards: 70
User-Agent: webrtc2sip Media Server 2.6.0

v=0
o=doubango 1983 678901 IN IP4 192.168.1.241
s=-
c=IN IP4 192.168.1.241
t=0 0
a=acap:1 setup:actpass
a=tcap:1 UDP/TLS/RTP/SAVPF UDP/TLS/RTP/SAVP RTP/SAVPF RTP/SAVP RTP/AVPF
a=acap:4 fingerprint:sha-1 CF:A9:24:08:BF:48:C5:D3:4F:84:69:1F:9B:A3:51:6A:E6:C1:B1:18
a=acap:3 fingerprint:sha-256 D8:2F:5E:63:F1:82:BA:F1:AE:57:AD:4A:39:CA:19:44:94:F0:D6:E6:49:73:6C:5F:E6:3E:57:59:9D:6A:B1:70
a=acap:2 connection:new
m=audio 48274 RTP/AVP 9 101
c=IN IP4 192.168.1.241
a=ptime:20
a=minptime:1
a=maxptime:255
a=silenceSupp:off - - - -
a=rtpmap:9 G722/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=acap:5 crypto:1 AES_CM_128_HMAC_SHA1_80 inline:rmMeibdry+kJVn2bNAcXRrOOq/MrSuts8++ENryL
a=acap:6 crypto:2 AES_CM_128_HMAC_SHA1_32 inline:8jUKeKlr3b7dclx930Iz/JgnF+hJF/TmzUDwySnt
a=pcfg:1 t=1 a=1,2,4|3
a=pcfg:2 t=2 a=1,2,4|3
a=pcfg:3 t=3 a=5,6
a=pcfg:4 t=4 a=5,6
a=pcfg:5 t=5
a=sendrecv
a=rtcp-mux
a=ssrc:1593906304 cname:673b17dabeb58611285200d107b4fe66
a=ssrc:1593906304 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:1593906304 label:doubango@audio
a=ice-ufrag:NkopHXZc5gY7TBI
a=ice-pwd:e9XKLEmPFpWJixoKAgydYf
a=candidate:TTEjOJzb60Ffvcj 1 udp 2130706431 192.168.1.241 48274 typ host tr udp fd 18
a=candidate:TTEjOJzb60Ffvcj 2 udp 2130706430 192.168.1.241 48275 typ host tr udp fd 19
a=candidate:HDnkDId4dP9cGVI 1 udp 2130706175 192.168.247.10 44986 typ host tr udp fd 20
a=candidate:HDnkDId4dP9cGVI 2 udp 2130706174 192.168.247.10 44987 typ host tr udp fd 21
a=candidate:srflxTTEjOJzb60 1 udp 1694498815 80.229.142.197 48274 typ srflx raddr 192.168.1.241 rport 48274 tr udp fd 18
a=candidate:srflxTTEjOJzb60 2 udp 1694498814 80.229.142.197 48275 typ srflx raddr 192.168.1.241 rport 48275 tr udp fd 19
<------------->
--- (11 headers 39 lines) ---
Sending to 192.168.1.241:10060 (no NAT)
Sending to 192.168.1.241:10060 (no NAT)
Using INVITE request as basis request - a66c6532-4453-2020-9839-3c44e616a9db
Found peer '60002' for '60002' from 192.168.1.241:10060

<--- Reliably Transmitting (no NAT) to 192.168.1.241:10060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.241:10060;branch=z9hG4bK-646837569;received=192.168.1.241;rport=10060
From: <sip:60002@192.168.1.241>;tag=2080760246
To: <sip:60007@192.168.1.241>;tag=as1359c37b
Call-ID: a66c6532-4453-2020-9839-3c44e616a9db
CSeq: 2017927504 INVITE
Server: Asterisk PBX 13.3.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="192.168.1.241", nonce="7f887154"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'a66c6532-4453-2020-9839-3c44e616a9db' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.241:10060 --->
ACK sip:60007@192.168.1.241 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.241:10060;branch=z9hG4bK-646837569;rport
From: <sip:60002@192.168.1.241>;tag=2080760246
To: <sip:60007@192.168.1.241>;tag=as1359c37b
Call-ID: a66c6532-4453-2020-9839-3c44e616a9db
CSeq: 2017927504 ACK
Content-Length: 0
Max-Forwards: 70

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

<--- SIP read from UDP:192.168.1.241:10060 --->
INVITE sip:60007@192.168.1.241 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.241:10060;branch=z9hG4bK-600636497;rport
From: <sip:60002@192.168.1.241>;tag=2080760246
To: <sip:60007@192.168.1.241>
Contact: <sip:60002@192.168.1.241:10060;ws-src-ip=192.168.1.4;ws-src-port=60457;ws-src-proto=ws;transport=udp>
Call-ID: a66c6532-4453-2020-9839-3c44e616a9db
CSeq: 2017927505 INVITE
Content-Type: application/sdp
Content-Length: 1753
Max-Forwards: 70
Authorization: Digest username="60002",realm="192.168.1.241",nonce="7f887154",uri="sip:60007@192.168.1.241",response="8f0ac0534260996bde582de180c95a11",algorithm=MD5
User-Agent: webrtc2sip Media Server 2.6.0

v=0
o=doubango 1983 678901 IN IP4 192.168.1.241
s=-
c=IN IP4 192.168.1.241
t=0 0
a=acap:1 setup:actpass
a=tcap:1 UDP/TLS/RTP/SAVPF UDP/TLS/RTP/SAVP RTP/SAVPF RTP/SAVP RTP/AVPF
a=acap:4 fingerprint:sha-1 CF:A9:24:08:BF:48:C5:D3:4F:84:69:1F:9B:A3:51:6A:E6:C1:B1:18
a=acap:3 fingerprint:sha-256 D8:2F:5E:63:F1:82:BA:F1:AE:57:AD:4A:39:CA:19:44:94:F0:D6:E6:49:73:6C:5F:E6:3E:57:59:9D:6A:B1:70
a=acap:2 connection:new
m=audio 48274 RTP/AVP 9 101
c=IN IP4 192.168.1.241
a=ptime:20
a=minptime:1
a=maxptime:255
a=silenceSupp:off - - - -
a=rtpmap:9 G722/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=acap:5 crypto:1 AES_CM_128_HMAC_SHA1_80 inline:rmMeibdry+kJVn2bNAcXRrOOq/MrSuts8++ENryL
a=acap:6 crypto:2 AES_CM_128_HMAC_SHA1_32 inline:8jUKeKlr3b7dclx930Iz/JgnF+hJF/TmzUDwySnt
a=pcfg:1 t=1 a=1,2,4|3
a=pcfg:2 t=2 a=1,2,4|3
a=pcfg:3 t=3 a=5,6
a=pcfg:4 t=4 a=5,6
a=pcfg:5 t=5
a=sendrecv
a=rtcp-mux
a=ssrc:1593906304 cname:673b17dabeb58611285200d107b4fe66
a=ssrc:1593906304 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:1593906304 label:doubango@audio
a=ice-ufrag:NkopHXZc5gY7TBI
a=ice-pwd:e9XKLEmPFpWJixoKAgydYf
a=candidate:TTEjOJzb60Ffvcj 1 udp 2130706431 192.168.1.241 48274 typ host tr udp fd 18
a=candidate:TTEjOJzb60Ffvcj 2 udp 2130706430 192.168.1.241 48275 typ host tr udp fd 19
a=candidate:HDnkDId4dP9cGVI 1 udp 2130706175 192.168.247.10 44986 typ host tr udp fd 20
a=candidate:HDnkDId4dP9cGVI 2 udp 2130706174 192.168.247.10 44987 typ host tr udp fd 21
a=candidate:srflxTTEjOJzb60 1 udp 1694498815 80.229.142.197 48274 typ srflx raddr 192.168.1.241 rport 48274 tr udp fd 18
a=candidate:srflxTTEjOJzb60 2 udp 1694498814 80.229.142.197 48275 typ srflx raddr 192.168.1.241 rport 48275 tr udp fd 19
<------------->
--- (12 headers 39 lines) ---
Sending to 192.168.1.241:10060 (no NAT)
Using INVITE request as basis request - a66c6532-4453-2020-9839-3c44e616a9db
Found peer '60002' for '60002' from 192.168.1.241:10060
Found RTP audio format 9
Found RTP audio format 101
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - (g722), peer - audio=(g722)/video=(nothing)/text=(nothing), combined - (g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.241:48274
Looking for 60007 in from-sip (domain 192.168.1.241)
sip_route_dump: route/path hop: <sip:60002@192.168.1.241:10060;ws-src-ip=192.168.1.4;ws-src-port=60457;ws-src-proto=ws;transport=udp>

<--- Transmitting (no NAT) to 192.168.1.241:10060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.241:10060;branch=z9hG4bK-600636497;received=192.168.1.241;rport=10060
From: <sip:60002@192.168.1.241>;tag=2080760246
To: <sip:60007@192.168.1.241>
Call-ID: a66c6532-4453-2020-9839-3c44e616a9db
CSeq: 2017927505 INVITE
Server: Asterisk PBX 13.3.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:60007@192.168.1.241:5060>
Content-Length: 0


<------------>
Really destroying SIP dialog '1419480c25e0d50e5af3538d3ceba49b@127.0.1.1:5060' Method: INVITE
[May 12 15:17:27] WARNING[10528][C-00000010]: app_dial.c:2431 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)

<--- Reliably Transmitting (no NAT) to 192.168.1.241:10060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.241:10060;branch=z9hG4bK-600636497;received=192.168.1.241;rport=10060
From: <sip:60002@192.168.1.241>;tag=2080760246
To: <sip:60007@192.168.1.241>;tag=as566107da
Call-ID: a66c6532-4453-2020-9839-3c44e616a9db
CSeq: 2017927505 INVITE
Server: Asterisk PBX 13.3.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-Asterisk-HangupCause: Subscriber absent
X-Asterisk-HangupCauseCode: 20
Content-Length: 0


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

<--- SIP read from UDP:192.168.1.241:10060 --->
ACK sip:60007@192.168.1.241 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.241:10060;branch=z9hG4bK-600636497;rport
From: <sip:60002@192.168.1.241>;tag=2080760246
To: <sip:60007@192.168.1.241>;tag=as566107da
Call-ID: a66c6532-4453-2020-9839-3c44e616a9db
CSeq: 2017927505 ACK
Content-Length: 0
Max-Forwards: 70

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'a66c6532-4453-2020-9839-3c44e616a9db' Method: ACK
Mjolnir*CLI> sip set debug off
SIP Debugging Disabled

sip.conf

 [general]
 realm=192.168.1.241
 udpbindaddr=192.168.1.241
 transport=udp;

 context=from-sip                        ; Default context for incoming calls
 ;
 bindport=5060                   ; bindport is the local UDP port that Asterisk will listen on
 bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
 ;
 ;disallow=all                    ; First disallow all codecs
 ;allow=g729
 disallow=all
 ;allow=g729
 ;allow=all
 allow=g722
 ;
 videosupport=yes;
tcpenable=yes;
transport=tcp,udp,ws,wss;
avpf=no ;
force_avp=no ; Force Asterisk to use avp. Introduced in Asterisk 11.11
dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS cert file is
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS private key is
dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when setting up DTLS
directmedia=no ; Asterisk will relay media for this peer
rtcachefriends=yes
encryption=no

ext 60002 (Firefox)

id: 1095 name: 60002 callerid: NULL defaultuser: 60002 regexten: NULL secret: password mailbox: NULL accountcode: NULL context: from-sip amaflags: NULL callgroup: NULL canreinvite: no defaultip: NULL dtmfmode: NULL fromuser: NULL fromdomain: NULL fullcontact: sip:60002@192.168.1.241:10060^3Brtcweb-breaker=yes^3Btransport=udp^3Bws-src-ip=1 host: dynamic insecure: NULL language: NULL md5secret: NULL nat: no deny: NULL permit: NULL mask: NULL pickupgroup: NULL port: 10060 qualify: NULL restrictcid: NULL rtptimeout: NULL rtpholdtimeout: NULL type: friend disallow: all allow: g722 musiconhold: NULL regseconds: 1430928012 ipaddr: 192.168.1.241 cancallforward: yes lastms: 0 useragent: regserver: guid: 8c981eca-4a2a-452c-b6a3-5af7ef360002 objecthash: ddc353f0382a6579b699b61e1fe99fe9 transport: udp avpf: no encryption: no icesupport: yes dtlsenable: no dtlsverify: no dtlscertfile: /etc/asterisk/keys/asterisk.pem dtlsprivatekey: /etc/asterisk/keys/asterisk.key dtlssetup: actpass directmedia: outgoing force_avp: NULL callbackextension: NULL

ext 60007 (SFLPhone)

id: 1100 name: 60007 callerid: NULL defaultuser: 60007 regexten: NULL secret: password mailbox: NULL accountcode: NULL context: from-sip amaflags: NULL callgroup: NULL canreinvite: no defaultip: NULL dtmfmode: NULL fromuser: NULL fromdomain: NULL fullcontact: sip:60007@192.168.1.59:5060 host: dynamic insecure: NULL language: NULL md5secret: NULL nat: no deny: NULL permit: NULL mask: NULL pickupgroup: NULL port: 5060 qualify: NULL restrictcid: NULL rtptimeout: NULL rtpholdtimeout: NULL type: friend disallow: all allow: g722 musiconhold: NULL regseconds: 1430926550 ipaddr: 192.168.1.59 cancallforward: yes lastms: 0 useragent: SFLphone/1.3.0 regserver: guid: 8c981eca-4a2a-452c-b6a3-5af7ef360007 objecthash: 74ed134c37b25570ec2f3a17f18823ed transport: udp,ws avpf: no encryption: no icesupport: no dtlsenable: yes dtlsverify: no dtlscertfile: /etc/asterisk/keys/asterisk.pem dtlsprivatekey: /etc/asterisk/keys/asterisk.key dtlssetup: actpass directmedia: no force_avp: NULL callbackextension: NULL

dialplan

mysql> select * from extensions; +-----+----------+-------+----------+------+-----------------+ | id | context | exten | priority | app | appdata | +-----+----------+-------+----------+------+-----------------+ | 257 | from-sip | _X. | 1 | Dial | SIP/${EXTEN},30 | +-----+----------+-------+----------+------+-----------------

charlesc · May 27, 2015, 1:20pm

Anyone got any ideas? I’ve been working on other stuff in the hope that someone can suggest something, but I need to start working on this again.

All I can think of doing is ripping out all the realtime stuff so I’ve got the most basic phone->asterisk->webrtc2sip case that other people have managed to get working, then try and figure out what’s different, but that’s going to be time consuming.

charlesc · June 3, 2015, 8:52am

Finally got to the bottom of this. It seems encryption=true in the global sip.conf overrides encryption=false in the sip_peers table. I swapped it round so the global setting is false, and then set it true where I need it in sip_peers. That seems to work.

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Asterisk realtime, webrtc2sip & SipML5

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