Unable to do SIP calling between WebRTC clients

Getting below error while installing libsrtp 1.5.4

[root@Nipun libsrtp-1.5.4]# make
make: AR@: Command not found
make: *** [libsrtp.a] Error 127

Did you read the documentation. Usually a file called README or install??
With all due respect if you are not able to install basic dependencies how do you expect to configure the WebRTC part on your linux server? If you are only copying and pasting commands you will not understand or get anything of this, either start reading more or hire someone to do the job.

1 Like

Hi,
We are getting the below error for incoming calls. Please help me.

sig_analog.c:2418 __analog_ss_thread: DTMFCID timed out waiting for ring. Exiting simple switch

r is this another thread? Seems an issue with your analog line or configuration.

@navaismo,
Thanks for the reply.

We are able to do outgoing calls. Please find the below configuration files.

extensions.conf

[from-pstn]

exten => s,1,Set(ODBC_SAVE_CDR()=${id},${clid},${src},${dst},${dcontext},${channel},${dstchannel},${lastapp},${lastdata},${start},${answer},${end},${duration},${billsec},${disposition},${amaflags},${accountcode},${uniqueid},${userfield},${sequence})

exten => s,n,Set(MONITOR_FILENAME=${STRFTIME(${EPOCH},%Y%m%d-%H%M%S)}-${CALLERID(num)})

exten => s,n,MixMonitor(/var/www/html/httpshare/record/incoming/${MONITOR_FILENAME}.wav,b)

exten => s,n,GoToIfTime(23:06-08:59|mon-sat||?fail)

exten => s,n,GoToIfTime(00:00-23:59|sun||?fail)

exten => s,n,GoToIfTime(09:00-23:05|mon-sat||?suc)

exten => s,n(suc),Dial(SIP/6002,20)

exten => s,n,Hangup

exten => s,n(fail),Answer

exten => s,n,Playback(./welcome)

exten => s,n,Hangup

sip.conf

;extension to use on web client
[6000]
host=dynamic
secret=6000
context=from-sip
type=friend
encryption=yes
avpf=yes
icesupport=yes
transport=ws,wss,udp
directmedia=no
disallow=all
dial = SIP/6000
allow=ulaw
allow=alaw
allow=sppex
allow=gsm
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
nat=yes,force_rport,comedia
;extension to use on softphones such as twinkle, linphone,ekiga…etc
[6002]
host=dynamic
secret=6002
context=from-sip
type=friend
transport=ws,wss,udp
directmedia=no
disallow=all
allow=all
nat=yes,force_rport

chan_dahdi.conf

[trunkgroups]

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no

cidstart=polarity_IN
cidsignalling=dtmf
callerid=asreceived
cid_rxgain=5.0

;Sangoma AFT-200 [slot:4 bus:4 span:1]
context=from-pstn
group=0
echocancel=yes
signalling = fxs_ks
channel => 1

wanpipe1.conf

[devices]
wanpipe1 = WAN_AFT_ANALOG, Comment

[interfaces]
w1g1 = wanpipe1, , TDM_VOICE, Comment

[wanpipe1]
CARD_TYPE = AFT
S514CPU = A
CommPort = PRI
AUTO_PCISLOT = NO
PCISLOT = 4
PCIBUS = 4
FE_MEDIA = FXO/FXS
TDMV_LAW = MULAW
TDMV_OPERMODE = FCC
RM_BATTTHRESH = 3
RM_BATTDEBOUNCE = 16
FE_NETWORK_SYNC = NO
MTU = 1500
UDPPORT = 9000
TTL = 255
IGNORE_FRONT_END = NO
TDMV_SPAN = 1
TE_AIS_MAINTENANCE = NO #NO: defualt YES: Start port in AIS Blue Alarm and keep line down
#wanpipemon -i w1g1 -c Ttx_ais_off to disable AIS maintenance mode
#wanpipemon -i w1g1 -c Ttx_ais_on to enable AIS maintenance mode
TDMV_HW_DTMF = NO # YES: receive dtmf events from hardware
TDMV_HW_FAX_DETECT = NO # YES: receive fax 1100hz events from hardware
HWEC_OPERATION_MODE = OCT_NORMAL # OCT_NORMAL: echo cancelation enabled with nlp (default)
# OCT_SPEECH: improves software tone detection by disabling NLP (echo possible)
# OCT_NO_ECHO:disables echo cancelation but allows VQE/tone functions.
HWEC_DTMF_REMOVAL = NO # NO: default YES: remove dtmf out of incoming media (must have hwdtmf enabled)
HWEC_NOISE_REDUCTION = NO # NO: default YES: reduces noise on the line - could break fax
HWEC_ACUSTIC_ECHO = NO # NO: default YES: enables acustic echo cancelation
HWEC_NLP_DISABLE = NO # NO: default YES: guarantees software tone detection (possible echo)
HWEC_TX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio level to be maintained (-20 default)
HWEC_RX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio level to be maintained (-20 default)
HWEC_TX_GAIN = 0 # 0: disable -24-24: db values to be applied to tx signal

RM_FAKE_POLARITY = YES
RM_FAKE_POLARITY_THRESHOLD = 16000
RM_FAKE_POLARITY_CIDTIMER = 10
RM_FAKE_POLARITY_CIDTIMEOUT = 4000
RM_FXOTXGAIN=1
RM_FXORXGAIN=1

RM_RING_DEBOUNCE = 0 #Value in ms: [0,24,32,54] [0 == 54]=Default, Set to lower value if initial ring is very short

[w1g1]
ACTIVE_CH = ALL
MTU = 8
TDMV_HWEC = YES

Contact Sangoma for Wanpipe support. In general you will not get analogue hardware support on this forum. Even for Digium hardware, you will get redirected to their commercial support arrangements, and competitor hardware has never, to my knowledge, produced a useful response.

Also, as has been pointed out, this should have been in a new thread, as it has absolutely nothing to do with the original problem.

1 Like

Hi all,

I am getting the below error for outgoing calls from webrtc client.

chan_sip.c:10837 process_sdp: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio

Please find the sip debug below

e[KNipun*CLI> sip set debug on

Nipun*CLI>
e[0KSIP Debugging enabled

e[KNipun*CLI>
e[0K
<— SIP read from UDP:188.165.231.30:11060 —>
ACK sip:09000957493@192.168.15.65 SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:11060;branch=z9hG4bK5BHArTHVmOKulwT2zKeZl9JwzXTei74A;rport
From: "Test"sip:6000@122.169.252.112;tag=A0JLTnSSfkJC46zCz0aN
To: sip:09000957493@122.169.252.112;tag=63532001117
Call-ID: 8e8d20cf-d56c-6e8a-e8f2-4cfb5e633fcd
CSeq: 33705 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP 183.83.32.3:51738;rport;branch=z9hG4bK5BHArTHVmOKulwT2zKeZl9JwzXTei74A;ws-hacked=WSS

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

e[KNipun*CLI>
e[0KReally destroying SIP dialog ‘43737432-fb89-3371-919b-fbf967e5ffc8’ Method: ACK

e[KNipun*CLI>
e[0K
<— SIP read from UDP:188.165.231.30:11060 —>
ACK sip:09000957493@192.168.15.65 SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:11060;branch=z9hG4bK5BHArTHVmOKulwT2zKeZl9JwzXTei74A;rport
From: "Test"sip:6000@122.169.252.112;tag=A0JLTnSSfkJC46zCz0aN
To: sip:09000957493@122.169.252.112;tag=63532001117
Call-ID: 8e8d20cf-d56c-6e8a-e8f2-4cfb5e633fcd
CSeq: 33705 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP 183.83.32.3:51738;rport;branch=z9hG4bK5BHArTHVmOKulwT2zKeZl9JwzXTei74A;ws-hacked=WSS
Via: SIP/2.0/TCP 183.83.32.3:51738;branch=z9hG4bK5BHArTHVmOKulwT2zKeZl9JwzXTei74A;rport;ws-hacked=WSS

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

e[KNipun*CLI>
e[0K
<— SIP read from UDP:188.165.231.30:11060 —>
INVITE sip:09000957493@192.168.15.65 SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:11060;branch=z9hG4bK-63795712975;rport
From: sip:6000@122.169.252.112;tag=63861741105
To: sip:09000957493@122.169.252.112
Contact: sip:6000@188.165.231.30:11060;ws-src-ip=183.83.32.3;ws-src-port=51738;ws-src-proto=wss;transport=udp
Call-ID: adb388cb-eaa5-feaf-6e06-c9dbeb7886ff
CSeq: 107759886 INVITE
Content-Type: application/sdp
Content-Length: 1480
Max-Forwards: 70
User-Agent: webrtc2sip Media Server 2.7.0

v=0
o=doubango 1983 678902 IN IP4 188.165.231.30
s=-
c=IN IP4 188.165.231.30
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 FA:CC:78:61:FC:3F:EE:CE:26:C9:2D:9D:0F:C7:40:52:50:B6:2D:F5
a=acap:3 fingerprint:sha-256 3D:51:2F:ED:79:D0:45:68:7D:6D:1A:9F:7D:49:D4:EF:C7:55:82:59:F7:D0:81:06:73:E7:55:87:C2:B1:DC:CF
a=acap:1 setup:actpass
m=audio 38660 RTP/AVP 111 8 0 101
c=IN IP4 188.165.231.30
a=ptime:20
a=minptime:1
a=maxptime:255
a=silenceSupp:off - - - -
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxplaybackrate=48000; sprop-maxcapturerate=48000; stereo=0; sprop-stereo=0; useinbandfec=0; usedtx=0
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/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:41Mxac1Hf0HIqx+PEneQX1CDosFfvYpwAXlvfZ1I
a=acap:6 crypto:2 AES_CM_128_HMAC_SHA1_32 inline:G6eWK9gTozy+RIVinTIMIqHufVfBRh7ocDGvYFR6
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:1355655309 cname:(null)
a=ssrc:1355655309 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:1355655309 label:doubango@audio
a=ice-ufrag:SX9e1BJn5pWoUxD
a=ice-pwd:fXeiZ191fmZFXvAhNn6uCt
a=candidate:FPyKb52TmtWDpks 1 udp 2130706431 188.165.231.30 38660 typ host tr udp fd 57
a=candidate:FPyKb52TmtWDpks 2 udp 2130706430 188.165.231.30 38661 typ host tr udp fd 58
<------------->
— (11 headers 38 lines) —

e[KNipun*CLI>
e[0KSending to 188.165.231.30:11060 (no NAT)
Sending to 188.165.231.30:11060 (no NAT)
Using INVITE request as basis request - adb388cb-eaa5-feaf-6e06-c9dbeb7886ff
Found peer ‘6000’ for ‘6000’ from 188.165.231.30:11060

e[KNipun*CLI>
e[0K
<— Reliably Transmitting (NAT) to 188.165.231.30:11060 —>
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 188.165.231.30:11060;branch=z9hG4bK-63795712975;received=188.165.231.30;rport=11060

From: sip:6000@122.169.252.112;tag=63861741105

To: sip:09000957493@122.169.252.112;tag=as2afe7f20

Call-ID: adb388cb-eaa5-feaf-6e06-c9dbeb7886ff

CSeq: 107759886 INVITE

Server: Asterisk PBX 14.5.0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm=“122.169.252.112”, nonce=“6ddea77d”

Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘adb388cb-eaa5-feaf-6e06-c9dbeb7886ff’ in 32000 ms (Method: INVITE)

e[KNipun*CLI>
e[0K
<— SIP read from UDP:188.165.231.30:11060 —>
ACK sip:09000957493@192.168.15.65 SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:11060;branch=z9hG4bK-63795712975;rport
From: sip:6000@122.169.252.112;tag=63861741105
To: sip:09000957493@122.169.252.112;tag=as2afe7f20
Call-ID: adb388cb-eaa5-feaf-6e06-c9dbeb7886ff
CSeq: 107759886 ACK
Content-Length: 0
Max-Forwards: 70

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

e[KNipun*CLI>
e[0K
<— SIP read from UDP:188.165.231.30:11060 —>
INVITE sip:09000957493@192.168.15.65 SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:11060;branch=z9hG4bK-63343420967;rport
From: sip:6000@122.169.252.112;tag=63861741105
To: sip:09000957493@122.169.252.112
Contact: sip:6000@188.165.231.30:11060;ws-src-ip=183.83.32.3;ws-src-port=51738;ws-src-proto=wss;transport=udp
Call-ID: adb388cb-eaa5-feaf-6e06-c9dbeb7886ff
CSeq: 107759887 INVITE
Content-Type: application/sdp
Content-Length: 1480
Max-Forwards: 70
Authorization: Digest username=“6000”,realm=“122.169.252.112”,nonce=“6ddea77d”,uri="sip:09000957493@122.169.252.112",response=“f1c500827da5c86bd730239cf2d13acf”,algorithm=MD5
User-Agent: webrtc2sip Media Server 2.7.0

v=0
o=doubango 1983 678902 IN IP4 188.165.231.30
s=-
c=IN IP4 188.165.231.30
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 FA:CC:78:61:FC:3F:EE:CE:26:C9:2D:9D:0F:C7:40:52:50:B6:2D:F5
a=acap:3 fingerprint:sha-256 3D:51:2F:ED:79:D0:45:68:7D:6D:1A:9F:7D:49:D4:EF:C7:55:82:59:F7:D0:81:06:73:E7:55:87:C2:B1:DC:CF
a=acap:1 setup:actpass
m=audio 38660 RTP/AVP 111 8 0 101
c=IN IP4 188.165.231.30
a=ptime:20
a=minptime:1
a=maxptime:255
a=silenceSupp:off - - - -
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxplaybackrate=48000; sprop-maxcapturerate=48000; stereo=0; sprop-stereo=0; useinbandfec=0; usedtx=0
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/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:41Mxac1Hf0HIqx+PEneQX1CDosFfvYpwAXlvfZ1I
a=acap:6 crypto:2 AES_CM_128_HMAC_SHA1_32 inline:G6eWK9gTozy+RIVinTIMIqHufVfBRh7ocDGvYFR6
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:1355655309 cname:(null)
a=ssrc:1355655309 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:1355655309 label:doubango@audio
a=ice-ufrag:SX9e1BJn5pWoUxD
a=ice-pwd:fXeiZ191fmZFXvAhNn6uCt
a=candidate:FPyKb52TmtWDpks 1 udp 2130706431 188.165.231.30 38660 typ host tr udp fd 57
a=candidate:FPyKb52TmtWDpks 2 udp 2130706430 188.165.231.30 38661 typ host tr udp fd 58
<------------->

e[KNipun*CLI>
e[0K— (12 headers 38 lines) —

e[KNipun*CLI>
e[0KSending to 188.165.231.30:11060 (NAT)

e[KNipun*CLI>
e[0KUsing INVITE request as basis request - adb388cb-eaa5-feaf-6e06-c9dbeb7886ff

e[KNipun*CLI>
e[0KFound peer ‘6000’ for ‘6000’ from 188.165.231.30:11060

e[KNipun*CLI>
e[0Ke[1;30m == e[0mDTLS ECDH initialized (secp256r1), faster PFS enabled

e[KNipun*CLI>
e[0Ke[1;30m == e[0mUsing SIP RTP CoS mark 5

e[KNipun*CLI>
e[0KFound RTP audio format 111

e[KNipun*CLI>
e[0KFound RTP audio format 8

e[KNipun*CLI>
e[0KFound RTP audio format 0

e[KNipun*CLI>
e[0KFound RTP audio format 101

e[KNipun*CLI>
e[0KFound audio description format opus for ID 111
Found audio description format PCMA for ID 8

e[KNipun*CLI>
e[0KFound audio description format PCMU for ID 0

e[KNipun*CLI>
e[0KFound audio description format telephone-event for ID 101

e[KNipun*CLI>
e[0K[Jun 1 10:00:22] e[1;31mWARNINGe[0m[4019][C-000000bb]: e[1;37mchan_sip.ce[0m:e[1;37m10837e[0m e[1;37mprocess_sdpe[0m: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio

e[KNipun*CLI>
e[0K
<— Reliably Transmitting (NAT) to 188.165.231.30:11060 —>
SIP/2.0 488 Not acceptable here

Via: SIP/2.0/UDP 188.165.231.30:11060;branch=z9hG4bK-63343420967;received=188.165.231.30;rport=11060

From: sip:6000@122.169.252.112;tag=63861741105

To: sip:09000957493@122.169.252.112;tag=as2afe7f20

Call-ID: adb388cb-eaa5-feaf-6e06-c9dbeb7886ff

CSeq: 107759887 INVITE

Server: Asterisk PBX 14.5.0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘adb388cb-eaa5-feaf-6e06-c9dbeb7886ff’ in 32000 ms (Method: INVITE)

e[KNipun*CLI>
e[0K
<— SIP read from UDP:188.165.231.30:11060 —>
ACK sip:09000957493@192.168.15.65 SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:11060;branch=z9hG4bK-63343420967;rport
From: sip:6000@122.169.252.112;tag=63861741105
To: sip:09000957493@122.169.252.112;tag=as2afe7f20
Call-ID: adb388cb-eaa5-feaf-6e06-c9dbeb7886ff
CSeq: 107759887 ACK
Content-Length: 0
Max-Forwards: 70

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

e[KNipun*CLI>
e[0K
<— SIP read from UDP:188.165.231.30:11060 —>
ACK sip:09000957493@192.168.15.65 SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:11060;branch=z9hG4bKVqcn7U1CPc6K5uJL72zlHSeTPQek9wGq;rport
From: "Test"sip:6000@122.169.252.112;tag=8AgDfot4twChPgdFgZrm
To: sip:09000957493@122.169.252.112;tag=63685214371
Call-ID: b236455e-26ef-c76a-6cdc-f4e674bc3f8e
CSeq: 32030 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP 183.83.32.3:51738;rport;branch=z9hG4bKVqcn7U1CPc6K5uJL72zlHSeTPQek9wGq;ws-hacked=WSS

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

e[KNipun*CLI>
e[0K
<— SIP read from UDP:188.165.231.30:11060 —>
ACK sip:09000957493@192.168.15.65 SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:11060;branch=z9hG4bKVqcn7U1CPc6K5uJL72zlHSeTPQek9wGq;rport
From: "Test"sip:6000@122.169.252.112;tag=8AgDfot4twChPgdFgZrm
To: sip:09000957493@122.169.252.112;tag=63685214371
Call-ID: b236455e-26ef-c76a-6cdc-f4e674bc3f8e
CSeq: 32030 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP 183.83.32.3:51738;rport;branch=z9hG4bKVqcn7U1CPc6K5uJL72zlHSeTPQek9wGq;ws-hacked=WSS

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

e[KNipun*CLI>
e[0K
<— SIP read from UDP:188.165.231.30:11060 —>
ACK sip:09000957493@192.168.15.65 SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:11060;branch=z9hG4bKVqcn7U1CPc6K5uJL72zlHSeTPQek9wGq;rport
From: "Test"sip:6000@122.169.252.112;tag=8AgDfot4twChPgdFgZrm
To: sip:09000957493@122.169.252.112;tag=63685214371
Call-ID: b236455e-26ef-c76a-6cdc-f4e674bc3f8e
CSeq: 32030 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP 183.83.32.3:51738;rport;branch=z9hG4bKVqcn7U1CPc6K5uJL72zlHSeTPQek9wGq;ws-hacked=WSS

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

e[KNipun*CLI>
e[0K
<— SIP read from UDP:188.165.231.30:11060 —>
ACK sip:09000957493@192.168.15.65 SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:11060;branch=z9hG4bKVqcn7U1CPc6K5uJL72zlHSeTPQek9wGq;rport
From: "Test"sip:6000@122.169.252.112;tag=8AgDfot4twChPgdFgZrm
To: sip:09000957493@122.169.252.112;tag=63685214371
Call-ID: b236455e-26ef-c76a-6cdc-f4e674bc3f8e
CSeq: 32030 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP 183.83.32.3:51738;rport;branch=z9hG4bKVqcn7U1CPc6K5uJL72zlHSeTPQek9wGq;ws-hacked=WSS

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

e[KNipun*CLI>
e[0K
<— SIP read from UDP:188.165.231.30:11060 —>
ACK sip:09000957493@192.168.15.65 SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:11060;branch=z9hG4bKVqcn7U1CPc6K5uJL72zlHSeTPQek9wGq;rport
From: "Test"sip:6000@122.169.252.112;tag=8AgDfot4twChPgdFgZrm
To: sip:09000957493@122.169.252.112;tag=63685214371
Call-ID: b236455e-26ef-c76a-6cdc-f4e674bc3f8e
CSeq: 32030 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP 183.83.32.3:51738;rport;branch=z9hG4bKVqcn7U1CPc6K5uJL72zlHSeTPQek9wGq;ws-hacked=WSS

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

e[KNipun*CLI>
e[0K
<— SIP read from UDP:188.165.231.30:11060 —>
ACK sip:09000957493@192.168.15.65 SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:11060;branch=z9hG4bKVqcn7U1CPc6K5uJL72zlHSeTPQek9wGq;rport
From: "Test"sip:6000@122.169.252.112;tag=8AgDfot4twChPgdFgZrm
To: sip:09000957493@122.169.252.112;tag=63685214371
Call-ID: b236455e-26ef-c76a-6cdc-f4e674bc3f8e
CSeq: 32030 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP 183.83.32.3:51738;rport;branch=z9hG4bKVqcn7U1CPc6K5uJL72zlHSeTPQek9wGq;ws-hacked=WSS

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

e[KNipun*CLI> sip set debug one[Kff
e[0K
<— SIP read from UDP:188.165.231.30:11060 —>
ACK sip:09000957493@192.168.15.65 SIP/2.0
Via: SIP/2.0/UDP 188.165.231.30:11060;branch=z9hG4bKVqcn7U1CPc6K5uJL72zlHSeTPQek9wGq;rport
From: "Test"sip:6000@122.169.252.112;tag=8AgDfot4twChPgdFgZrm
To: sip:09000957493@122.169.252.112;tag=63685214371
Call-ID: b236455e-26ef-c76a-6cdc-f4e674bc3f8e
CSeq: 32030 ACK
Content-Length: 0
Max-Forwards: 70
Via: SIP/2.0/TCP 183.83.32.3:51738;rport;branch=z9hG4bKVqcn7U1CPc6K5uJL72zlHSeTPQek9wGq;ws-hacked=WSS

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

Please find my sip.conf

[general]
context=public
realm=X.X.X.X(My public IP)
udpbindaddr=0.0.0.0:5060

rtpbindaddr=X.X.X.X(server IP)

tcpenable=yes ; Enable server for incoming TCP connections (default is no)
tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)

tlsenable=yes ; Enable server for incoming TLS (secure) connections (default is no)
tlsbindaddr=0.0.0.0

websocket_enabled = true ; Set to false to prevent chan_sip from listening to websockets. This
; is neeeded when using chan_sip and res_pjsip_transport_websockets on
; the same system.

websocket_write_timeout = 100 ; Default write timeout to set on websocket transports.
; This value may need to be adjusted for connections where
; Asterisk must write a substantial amount of data and the
; receiving clients are slow to process the received information.
; Value is in milliseconds; default is 100 ms.

transport=udp,ws,wss ; Set the default transports. The order determines the primary default transport.
; If tcpenable=no and the transport set is tcp, we will fallback to UDP.

srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
; Specifying a port in a SIP peer definition or
; when dialing outbound calls will supress SRV
; lookups for that peer or call.

dtmfmode = rfc2833

tlscertfile=/etc/asterisk/keys/asterisk.pem ; Certificate chain (*.pem format only) to use for TLS connections
; The certificates must be sorted starting with the subject’s certificate
; and followed by intermediate CA certificates if applicable. If the
; file name ends in _rsa, for example “asterisk_rsa.pem”, the files
; “asterisk_dsa.pem” and/or “asterisk_ecc.pem” are loaded
; (certificate, intermediates, private key), to support multiple
; algorithms for server authentication (RSA, DSA, ECDSA). If the chains
; are different, at least OpenSSL 1.0.2 is required.
; Default is to look for “asterisk.pem” in current directory

tlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Private key file (*.pem format only) for TLS connections.
; If no tlsprivatekey is specified, tlscertfile is searched for
; for both public and private key.

rtcp_mux=yes

dtlsenable = yes ; Enable or disable DTLS-SRTP support
dtlsverify = yes

;extension to use on web client
[6000]
host=dynamic
secret=6000
context=from-sip
type=friend
encryption=yes
avpf=no
icesupport=yes
transport=ws,wss,udp
directmedia=no
disallow=all
dial = SIP/6000
allow=ulaw
allow=alaw
allow=sppex
allow=gsm
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
nat=force_rport,comedia
;extension to use on softphones such as twinkle, linphone,ekiga…etc
[6002]
host=dynamic
secret=6002
context=from-sip
type=friend
transport=ws,wss,udp
directmedia=no
disallow=all
allow=all
nat=yes,force_rport

Hi all,

Please let me know, is pjproject is mandatory for webrtc.

That SDP is using SDP capabilities[1] which are not supported by chan_sip or chan_pjsip, so to them there would be no crypto. And yes pjproject is required for WebRTC use in Asterisk as it provides the ICE support, which is mandatory in WebRTC.

[1] https://tools.ietf.org/html/rfc5939

Thank you for the reply.

Can you please tell me where I have to give no crypto in chan_sip.

@jcolp

We are getting below error while installing pjproject

g++ -c -fomit-frame-pointer -Wall -DPJ_AUTOCONF=1 -O2 -DNDEBUG -DPJ_IS_BIG_ENDIAN=0 -DPJ_IS_LITTLE_ENDIAN=1 -fPIC -I. -I…/…/yuv/include -I…/…/…/pjlib/include
-o output/libyuv-i686-pc-linux-gnu/row_common.o
…/…/yuv/source/row_common.cc
…/…/yuv/source/row_common.cc: In function ‘void libyuv::YuvPixel(uint8, uint8, uint8, uint8*, uint8*, uint8*, const libyuv::YuvConstants*)’:
…/…/yuv/source/row_common.cc:1256: error: invalid types ‘const signed char vector[int]’ for array subscript
…/…/yuv/source/row_common.cc:1257: error: invalid types ‘const signed char vector[int]’ for array subscript
…/…/yuv/source/row_common.cc:1258: error: invalid types ‘const signed char vector[int]’ for array subscript
…/…/yuv/source/row_common.cc:1259: error: invalid types ‘const signed char vector[int]’ for array subscript
…/…/yuv/source/row_common.cc:1260: error: invalid types ‘const short int vector[int]’ for array subscript
…/…/yuv/source/row_common.cc:1261: error: invalid types ‘const short int vector[int]’ for array subscript
…/…/yuv/source/row_common.cc:1262: error: invalid types ‘const short int vector[int]’ for array subscript
…/…/yuv/source/row_common.cc:1263: error: invalid types ‘const short int vector[int]’ for array subscript
make[3]: *** [output/libyuv-i686-pc-linux-gnu/row_common.o] Error 1
make[3]: Leaving directory /usr/src/pjproject-2.6/third_party/build/yuv' make[2]: *** [libyuv-i686-pc-linux-gnu.a] Error 2 make[2]: Leaving directory/usr/src/pjproject-2.6/third_party/build/yuv’
make[1]: *** [all] Error 1
make[1]: Leaving directory `/usr/src/pjproject-2.6/third_party/build’
make: *** [all] Error 1

I have resolved that error.

Now I am getting the below errors.

…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c: At top level:
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:221: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘mm_pow_ps’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c: In function ‘OverdriveAndSuppressSSE2’:
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:362: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_hNlFb’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:362: error: ‘vec_hNlFb’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:363: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_one’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:363: error: ‘vec_one’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:364: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_minus_one’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:364: error: ‘vec_minus_one’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:365: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_overDriveSm’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:365: error: ‘vec_overDriveSm’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:369: error: ‘__m128’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:369: error: expected ‘;’ before ‘vec_hNl’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:370: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_weightCurve’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:370: error: ‘vec_weightCurve’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:371: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘bigger’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:371: error: ‘bigger’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:371: error: ‘vec_hNl’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:372: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_weightCurve_hNlFb’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:372: error: ‘vec_weightCurve_hNlFb’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:373: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_one_weightCurve’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:373: error: ‘vec_one_weightCurve’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:374: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_one_weightCurve_hNl’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:374: error: ‘vec_one_weightCurve_hNl’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:376: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_if0’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:376: error: ‘vec_if0’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:377: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_if1’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:377: error: ‘vec_if1’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:382: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_overDriveCurve’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:382: error: ‘vec_overDriveCurve’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:384: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_overDriveSm_overDriveCurve’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:384: error: ‘vec_overDriveSm_overDriveCurve’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:386: warning: implicit declaration of function ‘mm_pow_ps’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:392: error: expected ‘;’ before ‘vec_efw_re’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:393: error: expected ‘;’ before ‘vec_efw_im’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:394: error: ‘vec_efw_re’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:395: error: ‘vec_efw_im’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c: At top level:
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:423: error: expected ‘)’ before ‘sum’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c: In function ‘PartitionDelay’:
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:443: error: ‘__m128’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:443: error: expected ‘;’ before ‘vec_wfEn’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:446: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_wfBuf0’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:446: error: ‘vec_wfBuf0’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:447: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_wfBuf1’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:447: error: ‘vec_wfBuf1’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:448: error: ‘vec_wfEn’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:451: warning: implicit declaration of function ‘_mm_add_ps_4x1’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c: In function ‘SmoothedPSD’:
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:486: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_15’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:486: error: ‘vec_15’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:487: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_GCoh0’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:487: error: ‘vec_GCoh0’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:488: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_GCoh1’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:488: error: ‘vec_GCoh1’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:489: error: ‘__m128’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:489: error: expected ‘;’ before ‘vec_sdSum’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:490: error: expected ‘;’ before ‘vec_seSum’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:493: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_dfw0’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:493: error: ‘vec_dfw0’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:494: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_dfw1’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:494: error: ‘vec_dfw1’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:495: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_efw0’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:495: error: ‘vec_efw0’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:496: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_efw1’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:496: error: ‘vec_efw1’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:497: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_xfw0’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:497: error: ‘vec_xfw0’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:498: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_xfw1’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:498: error: ‘vec_xfw1’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:499: error: expected ‘;’ before ‘vec_sd’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:500: error: expected ‘;’ before ‘vec_se’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:501: error: expected ‘;’ before ‘vec_sx’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:502: error: expected ‘;’ before ‘vec_dfw_sumsq’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:503: error: expected ‘;’ before ‘vec_efw_sumsq’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:504: error: expected ‘;’ before ‘vec_xfw_sumsq’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:505: error: ‘vec_dfw_sumsq’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:506: error: ‘vec_efw_sumsq’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:507: error: ‘vec_xfw_sumsq’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:508: warning: implicit declaration of function ‘_mm_max_ps’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:509: error: ‘vec_sd’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:510: error: ‘vec_se’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:511: error: ‘vec_sx’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:517: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_3210’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:517: error: ‘vec_3210’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:518: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_7654’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:518: error: ‘vec_7654’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:519: error: expected ‘;’ before ‘vec_a’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:521: error: expected ‘;’ before ‘vec_b’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:523: error: expected ‘;’ before ‘vec_dfwefw0011’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:524: error: expected ‘;’ before ‘vec_dfwefw0110’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:525: error: ‘vec_a’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:526: error: ‘vec_b’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:527: error: ‘vec_dfwefw0011’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:529: error: ‘vec_dfwefw0110’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:538: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_3210’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:539: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_7654’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:540: error: expected ‘;’ before ‘vec_a’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:542: error: expected ‘;’ before ‘vec_b’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:544: error: expected ‘;’ before ‘vec_dfwxfw0011’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:545: error: expected ‘;’ before ‘vec_dfwxfw0110’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:548: error: ‘vec_dfwxfw0011’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:550: error: ‘vec_dfwxfw0110’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:558: error: ‘vec_sdSum’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:559: error: ‘vec_seSum’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c: In function ‘WindowData’:
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:613: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_Buf1’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:613: error: ‘vec_Buf1’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:614: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_Buf2’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:614: error: ‘vec_Buf2’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:615: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_sqrtHanning’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:615: error: ‘vec_sqrtHanning’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:615: warning: implicit declaration of function ‘_mm_load_ps’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:617: error: ‘__m128’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:617: error: expected ‘;’ before ‘vec_sqrtHanning_rev’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:620: error: ‘vec_sqrtHanning_rev’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c: In function ‘StoreAsComplex’:
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:634: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_fft0’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:634: error: ‘vec_fft0’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:635: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_fft4’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:635: error: ‘vec_fft4’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:636: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_a’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:636: error: ‘vec_a’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:638: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_b’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:638: error: ‘vec_b’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c: In function ‘SubbandCoherenceSSE2’:
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:680: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_1eminus10’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:680: error: ‘vec_1eminus10’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:684: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_sd’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:684: error: ‘vec_sd’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:685: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_se’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:685: error: ‘vec_se’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:686: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_sx’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:686: error: ‘vec_sx’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:687: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_sdse’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:687: error: ‘vec_sdse’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:689: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_sdsx’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:689: error: ‘vec_sdsx’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:691: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_sde_3210’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:691: error: ‘vec_sde_3210’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:692: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_sde_7654’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:692: error: ‘vec_sde_7654’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:693: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_sxd_3210’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:693: error: ‘vec_sxd_3210’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:694: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_sxd_7654’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:694: error: ‘vec_sxd_7654’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:695: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_sde_0’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:695: error: ‘vec_sde_0’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:697: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_sde_1’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:697: error: ‘vec_sde_1’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:699: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_sxd_0’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:699: error: ‘vec_sxd_0’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:701: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘attribute’ before ‘vec_sxd_1’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:701: error: ‘vec_sxd_1’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:703: error: ‘__m128’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:703: error: expected ‘;’ before ‘vec_cohde’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:704: error: expected ‘;’ before ‘vec_cohxd’
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:705: error: ‘vec_cohde’ undeclared (first use in this function)
…/…/webrtc/src/webrtc//modules/audio_processing/aec/aec_core_sse2.c:707: error: ‘vec_cohxd’ undeclared (first use in this function)
make[3]: *** [output/libwebrtc-i686-pc-linux-gnu/modules/audio_processing/aec/aec_core_sse2.o] Error 1
make[3]: Leaving directory /usr/src/pjproject-2.6/third_party/build/webrtc' make[2]: *** [libwebrtc-i686-pc-linux-gnu.a] Error 2 make[2]: Leaving directory/usr/src/pjproject-2.6/third_party/build/webrtc’
make[1]: *** [all] Error 1
make[1]: Leaving directory `/usr/src/pjproject-2.6/third_party/build’
make: *** [all] Error 1

The recommended approach for building pjproject is to use the bundled support[1] in Asterisk. It automatically builds it with the correct configuration and disables anything not needed.

[1 ]http://blogs.asterisk.org/2016/03/16/asterisk-13-8-0-now-easier-pjsip-install-method/

@jcolp

With that bundled support also we are getting above errors.

I haven’t seen anyone else report such problems, so it is likely specific to your environment - be it installed packages or versions. I don’t really have a suggestion on how to rectify that.

@jcolp

Can you please tell me where I have to give no crypto in chan_sip.

According to the sip.conf.sample it is controlled using the “encryption” option.

@jcolp

I have given encryption=no for the sip client but we are getting the same error.

Which error? The compilation error or the SIP debug SDP error? You are using WebRTC2SIP media gateway, that gateway must be configured to use encryption, certificates too, so check that in your setup.

@navaismo

Thanks for the reply.

For that I need to give encryption=yes and below is my webrtc client configuration in sip.conf.

[6000]
host=dynamic
secret=6000
context=from-sip
type=friend
encryption=yes
avpf=no
icesupport=yes
transport=ws,wss,udp
directmedia=no
disallow=all
dial = SIP/6000
allow=ulaw
allow=alaw
allow=speex
allow=gsm
;allow=opus
;allow=vp8
dtlsenable=yes
dtlsverify=no
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
rtcp_mux=yes
dtmfmode=rfc2833
nat=force_rport,comedia

Please correct me, if anything required for webrtc2sip media gateway.