One-way audio on outgoing calls on Opus

Dear all,

I experience one-way audio (remote party can’t hear) using Opus codec on Asterisk 14.1.2. Other codecs don’t have this issue.

Please find the SIP logs below:

<--- Received SIP request (899 bytes) from UDP:197.225.152.105:61429 --->
INVITE sip:0170809700@voip.cahri.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:61429;branch=z9hG4bK-524287-1---fc65866454f75309;rport;alias
Max-Forwards: 70
Contact: <sip:julien@197.225.152.105:61429;rinstance=8c06fb3fbe8c1a2c>
To: <sip:0170809700@voip.cahri.net>
From: <sip:julien@voip.cahri.net>;tag=e48f0c01
Call-ID: ZThhNGI0MTk4YmY4MWQ3OWMwOTkzOTg2MzMxMTQ0ZTk
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, OPTIONS, UPDATE, PRACK, MESSAGE, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, 100rel
User-Agent: Bria iOS release 3.8.1 stamp 36276.36277
Content-Length: 282

v=0
o=- 1479226443496974 1 IN IP4 192.168.1.2
s=Cpc session
c=IN IP4 192.168.1.2
t=0 0
m=audio 53932 RTP/AVP 120 9 0 8 101
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<--- Transmitting SIP response (541 bytes) to UDP:197.225.152.105:61429 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:61429;rport=61429;received=197.225.152.105;branch=z9hG4bK-524287-1---fc65866454f75309;alias
Call-ID: ZThhNGI0MTk4YmY4MWQ3OWMwOTkzOTg2MzMxMTQ0ZTk
From: <sip:julien@voip.cahri.net>;tag=e48f0c01
To: <sip:0170809700@voip.cahri.net>;tag=z9hG4bK-524287-1---fc65866454f75309
CSeq: 1 INVITE
WWW-Authenticate: Digest  realm="asterisk",nonce="1479226444/9439e99a804765a02cb705211c16abe9",opaque="06a6ba4912dab6ac",algorithm=md5,qop="auth"
Server: Asterisk PBX 14.1.2
Content-Length:  0


<--- Received SIP request (899 bytes) from UDP:197.225.152.105:61429 --->
INVITE sip:0170809700@voip.cahri.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:61429;branch=z9hG4bK-524287-1---fc65866454f75309;rport;alias
Max-Forwards: 70
Contact: <sip:julien@197.225.152.105:61429;rinstance=8c06fb3fbe8c1a2c>
To: <sip:0170809700@voip.cahri.net>
From: <sip:julien@voip.cahri.net>;tag=e48f0c01
Call-ID: ZThhNGI0MTk4YmY4MWQ3OWMwOTkzOTg2MzMxMTQ0ZTk
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, OPTIONS, UPDATE, PRACK, MESSAGE, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, 100rel
User-Agent: Bria iOS release 3.8.1 stamp 36276.36277
Content-Length: 282

v=0
o=- 1479226443496974 1 IN IP4 192.168.1.2
s=Cpc session
c=IN IP4 192.168.1.2
t=0 0
m=audio 53932 RTP/AVP 120 9 0 8 101
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<--- Transmitting SIP response (541 bytes) to UDP:197.225.152.105:61429 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:61429;rport=61429;received=197.225.152.105;branch=z9hG4bK-524287-1---fc65866454f75309;alias
Call-ID: ZThhNGI0MTk4YmY4MWQ3OWMwOTkzOTg2MzMxMTQ0ZTk
From: <sip:julien@voip.cahri.net>;tag=e48f0c01
To: <sip:0170809700@voip.cahri.net>;tag=z9hG4bK-524287-1---fc65866454f75309
CSeq: 1 INVITE
WWW-Authenticate: Digest  realm="asterisk",nonce="1479226445/212fb2603c5f5ffd45db8c4d88e6f0dd",opaque="62db1a5c1948d0c0",algorithm=md5,qop="auth"
Server: Asterisk PBX 14.1.2
Content-Length:  0


<--- Received SIP request (365 bytes) from UDP:197.225.152.105:61429 --->
ACK sip:0170809700@voip.cahri.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:61429;branch=z9hG4bK-524287-1---fc65866454f75309;rport;alias
Max-Forwards: 70
To: <sip:0170809700@voip.cahri.net>;tag=z9hG4bK-524287-1---fc65866454f75309
From: <sip:julien@voip.cahri.net>;tag=e48f0c01
Call-ID: ZThhNGI0MTk4YmY4MWQ3OWMwOTkzOTg2MzMxMTQ0ZTk
CSeq: 1 ACK
Content-Length: 0


<--- Received SIP request (1192 bytes) from UDP:197.225.152.105:61429 --->
INVITE sip:0170809700@voip.cahri.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:61429;branch=z9hG4bK-524287-1---db525e6f7125285c;rport;alias
Max-Forwards: 70
Contact: <sip:julien@197.225.152.105:61429;rinstance=8c06fb3fbe8c1a2c>
To: <sip:0170809700@voip.cahri.net>
From: <sip:julien@voip.cahri.net>;tag=e48f0c01
Call-ID: ZThhNGI0MTk4YmY4MWQ3OWMwOTkzOTg2MzMxMTQ0ZTk
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, OPTIONS, UPDATE, PRACK, MESSAGE, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, 100rel
User-Agent: Bria iOS release 3.8.1 stamp 36276.36277
Authorization: Digest username="julien",realm="asterisk",nonce="1479226444/9439e99a804765a02cb705211c16abe9",uri="sip:0170809700@voip.cahri.net",response="0ab4758c151ce3845741099aeec1fb01",cnonce="4eb6f78f898f734a4d50cd1c073b3bd6",nc=00000001,qop=auth,algorithm=md5,opaque="06a6ba4912dab6ac"
Content-Length: 282

v=0
o=- 1479226443496974 1 IN IP4 192.168.1.2
s=Cpc session
c=IN IP4 192.168.1.2
t=0 0
m=audio 53932 RTP/AVP 120 9 0 8 101
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

  == Using SIP RTP Audio TOS bits 184
<--- Transmitting SIP response (348 bytes) to UDP:197.225.152.105:61429 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:61429;rport=61429;received=197.225.152.105;branch=z9hG4bK-524287-1---db525e6f7125285c;alias
Call-ID: ZThhNGI0MTk4YmY4MWQ3OWMwOTkzOTg2MzMxMTQ0ZTk
From: <sip:julien@voip.cahri.net>;tag=e48f0c01
To: <sip:0170809700@voip.cahri.net>
CSeq: 2 INVITE
Server: Asterisk PBX 14.1.2
Content-Length:  0


    -- Executing [0170809700@from-internal:1] Goto("PJSIP/julien-00000006", "dial-sip,0170809700,1") in new stack
    -- Goto (dial-sip,0170809700,1)
    -- Executing [0170809700@dial-sip:1] AGI("PJSIP/julien-00000006", "outboundcli_plugandtel") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/outboundcli_plugandtel
outboundcli_plugandtel: Request: {agi_request:outboundcli_plugandtel,agi_channel:PJSIP/julien-00000006,agi_language:fr,agi_type:PJSIP,agi_uniqueid:1479226445.14,agi_version:14.1.2,agi_callerid:821,agi_calleridname:Julien
 outboundcli_plugandtel: Original Caller ID: 821
 outboundcli_plugandtel: DNID: 0170809700
 outboundcli_plugandtel: Plugandtel Caller ID: 0262787821
    -- <PJSIP/julien-00000006>AGI Script outboundcli_plugandtel completed, returning 0
    -- Executing [0170809700@dial-sip:2] Macro("PJSIP/julien-00000006", "dialout,IAX2/plugandtel/0170809700,,T") in new stack
    -- Executing [s@macro-dialout:1] Dial("PJSIP/julien-00000006", "IAX2/plugandtel/0170809700,,T") in new stack
    -- Called IAX2/plugandtel/0170809700
<--- Received SIP request (1192 bytes) from UDP:197.225.152.105:61429 --->
INVITE sip:0170809700@voip.cahri.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:61429;branch=z9hG4bK-524287-1---db525e6f7125285c;rport;alias
Max-Forwards: 70
Contact: <sip:julien@197.225.152.105:61429;rinstance=8c06fb3fbe8c1a2c>
To: <sip:0170809700@voip.cahri.net>
From: <sip:julien@voip.cahri.net>;tag=e48f0c01
Call-ID: ZThhNGI0MTk4YmY4MWQ3OWMwOTkzOTg2MzMxMTQ0ZTk
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, OPTIONS, UPDATE, PRACK, MESSAGE, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, 100rel
User-Agent: Bria iOS release 3.8.1 stamp 36276.36277
Authorization: Digest username="julien",realm="asterisk",nonce="1479226444/9439e99a804765a02cb705211c16abe9",uri="sip:0170809700@voip.cahri.net",response="0ab4758c151ce3845741099aeec1fb01",cnonce="4eb6f78f898f734a4d50cd1c073b3bd6",nc=00000001,qop=auth,algorithm=md5,opaque="06a6ba4912dab6ac"
Content-Length: 282

v=0
o=- 1479226443496974 1 IN IP4 192.168.1.2
s=Cpc session
c=IN IP4 192.168.1.2
t=0 0
m=audio 53932 RTP/AVP 120 9 0 8 101
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<--- Transmitting SIP response (348 bytes) to UDP:197.225.152.105:61429 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:61429;rport=61429;received=197.225.152.105;branch=z9hG4bK-524287-1---db525e6f7125285c;alias
Call-ID: ZThhNGI0MTk4YmY4MWQ3OWMwOTkzOTg2MzMxMTQ0ZTk
From: <sip:julien@voip.cahri.net>;tag=e48f0c01
To: <sip:0170809700@voip.cahri.net>
CSeq: 2 INVITE
Server: Asterisk PBX 14.1.2
Content-Length:  0


<--- Received SIP request (365 bytes) from UDP:197.225.152.105:61429 --->
ACK sip:0170809700@voip.cahri.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:61429;branch=z9hG4bK-524287-1---fc65866454f75309;rport;alias
Max-Forwards: 70
To: <sip:0170809700@voip.cahri.net>;tag=z9hG4bK-524287-1---fc65866454f75309
From: <sip:julien@voip.cahri.net>;tag=e48f0c01
Call-ID: ZThhNGI0MTk4YmY4MWQ3OWMwOTkzOTg2MzMxMTQ0ZTk
CSeq: 1 ACK
Content-Length: 0


    -- Call accepted by 193.202.111.220:4569 (format alaw)
    -- Format for call is (alaw)
    -- IAX2/plugandtel-15756 is ringing
<--- Transmitting SIP response (536 bytes) to UDP:197.225.152.105:61429 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.2:61429;rport=61429;received=197.225.152.105;branch=z9hG4bK-524287-1---db525e6f7125285c;alias
Call-ID: ZThhNGI0MTk4YmY4MWQ3OWMwOTkzOTg2MzMxMTQ0ZTk
From: <sip:julien@voip.cahri.net>;tag=e48f0c01
To: <sip:0170809700@voip.cahri.net>;tag=bc3a3d1d-a2eb-4338-aa6f-52e9dc3e9ac7
CSeq: 2 INVITE
Server: Asterisk PBX 14.1.2
Contact: <sip:41.213.137.68:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Content-Length:  0


    -- IAX2/plugandtel-15756 stopped sounds
    -- IAX2/plugandtel-15756 answered PJSIP/julien-00000006
<--- Transmitting SIP response (1122 bytes) to UDP:197.225.152.105:61429 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:61429;rport=61429;received=197.225.152.105;branch=z9hG4bK-524287-1---db525e6f7125285c;alias
Call-ID: ZThhNGI0MTk4YmY4MWQ3OWMwOTkzOTg2MzMxMTQ0ZTk
From: <sip:julien@voip.cahri.net>;tag=e48f0c01
To: <sip:0170809700@voip.cahri.net>;tag=bc3a3d1d-a2eb-4338-aa6f-52e9dc3e9ac7
CSeq: 2 INVITE
Server: Asterisk PBX 14.1.2
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Contact: <sip:41.213.137.68:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   509

v=0
o=- 1052048910 3 IN IP4 41.213.137.68
s=Asterisk
c=IN IP4 41.213.137.68
t=0 0
m=audio 10770 RTP/AVP 120 101
a=ice-ufrag:21c53ddc1b74ee706f6129fb34435f41
a=ice-pwd:654ecfa87b42dc4a465cfb274f34ec07
a=candidate:H29d58944 1 UDP 2130706431 41.213.137.68 10770 typ host
a=candidate:H29d58944 2 UDP 2130706430 41.213.137.68 10771 typ host
a=rtpmap:120 opus/48000/2
a=fmtp:120 maxaveragebitrate=64000;usedtx=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

    -- Channel IAX2/plugandtel-15756 joined 'simple_bridge' basic-bridge <f292e1f7-f316-46f6-94ae-c838f0360bfc>
    -- Channel PJSIP/julien-00000006 joined 'simple_bridge' basic-bridge <f292e1f7-f316-46f6-94ae-c838f0360bfc>
<--- Transmitting SIP response (1122 bytes) to UDP:197.225.152.105:61429 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:61429;rport=61429;received=197.225.152.105;branch=z9hG4bK-524287-1---db525e6f7125285c;alias
Call-ID: ZThhNGI0MTk4YmY4MWQ3OWMwOTkzOTg2MzMxMTQ0ZTk
From: <sip:julien@voip.cahri.net>;tag=e48f0c01
To: <sip:0170809700@voip.cahri.net>;tag=bc3a3d1d-a2eb-4338-aa6f-52e9dc3e9ac7
CSeq: 2 INVITE
Server: Asterisk PBX 14.1.2
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Contact: <sip:41.213.137.68:5060>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   509

v=0
o=- 1052048910 3 IN IP4 41.213.137.68
s=Asterisk
c=IN IP4 41.213.137.68
t=0 0
m=audio 10770 RTP/AVP 120 101
a=ice-ufrag:21c53ddc1b74ee706f6129fb34435f41
a=ice-pwd:654ecfa87b42dc4a465cfb274f34ec07
a=candidate:H29d58944 1 UDP 2130706431 41.213.137.68 10770 typ host
a=candidate:H29d58944 2 UDP 2130706430 41.213.137.68 10771 typ host
a=rtpmap:120 opus/48000/2
a=fmtp:120 maxaveragebitrate=64000;usedtx=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:20
a=sendrecv

<--- Received SIP request (485 bytes) from UDP:197.225.152.105:61429 --->
ACK sip:41.213.137.68:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:61429;branch=z9hG4bK-524287-1---09b0e1072d018947;rport;alias
Max-Forwards: 70
Contact: <sip:julien@197.225.152.105:61429;rinstance=8c06fb3fbe8c1a2c>
To: <sip:0170809700@voip.cahri.net>;tag=bc3a3d1d-a2eb-4338-aa6f-52e9dc3e9ac7
From: <sip:julien@voip.cahri.net>;tag=e48f0c01
Call-ID: ZThhNGI0MTk4YmY4MWQ3OWMwOTkzOTg2MzMxMTQ0ZTk
CSeq: 2 ACK
User-Agent: Bria iOS release 3.8.1 stamp 36276.36277
Content-Length: 0


       > 0x7fe52401d080 -- Probation passed - setting RTP source address to 197.225.152.105:53932
       > 0x7fe52401d080 -- Probation passed - setting RTP source address to 197.225.152.105:53932
<--- Received SIP request (485 bytes) from UDP:197.225.152.105:61429 --->
ACK sip:41.213.137.68:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:61429;branch=z9hG4bK-524287-1---09b0e1072d018947;rport;alias
Max-Forwards: 70
Contact: <sip:julien@197.225.152.105:61429;rinstance=8c06fb3fbe8c1a2c>
To: <sip:0170809700@voip.cahri.net>;tag=bc3a3d1d-a2eb-4338-aa6f-52e9dc3e9ac7
From: <sip:julien@voip.cahri.net>;tag=e48f0c01
Call-ID: ZThhNGI0MTk4YmY4MWQ3OWMwOTkzOTg2MzMxMTQ0ZTk
CSeq: 2 ACK
User-Agent: Bria iOS release 3.8.1 stamp 36276.36277
Content-Length: 0


<--- Received SIP request (771 bytes) from UDP:197.225.152.105:61429 --->
BYE sip:41.213.137.68:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:61429;branch=z9hG4bK-524287-1---46169407acf8c62a;rport;alias
Max-Forwards: 70
Contact: <sip:julien@197.225.152.105:61429;rinstance=8c06fb3fbe8c1a2c>
To: <sip:0170809700@voip.cahri.net>;tag=bc3a3d1d-a2eb-4338-aa6f-52e9dc3e9ac7
From: <sip:julien@voip.cahri.net>;tag=e48f0c01
Call-ID: ZThhNGI0MTk4YmY4MWQ3OWMwOTkzOTg2MzMxMTQ0ZTk
CSeq: 3 BYE
User-Agent: Bria iOS release 3.8.1 stamp 36276.36277
Authorization: Digest username="julien",realm="asterisk",nonce="1479226444/9439e99a804765a02cb705211c16abe9",uri="sip:41.213.137.68:5060",response="a369b6a39720aec236e3e08b65d53566",cnonce="9e2016bc8386af5f437e726a66978d74",nc=00000002,qop=auth,algorithm=md5,opaque="06a6ba4912dab6ac"
Content-Length: 0


<--- Transmitting SIP response (382 bytes) to UDP:197.225.152.105:61429 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:61429;rport=61429;received=197.225.152.105;branch=z9hG4bK-524287-1---46169407acf8c62a;alias
Call-ID: ZThhNGI0MTk4YmY4MWQ3OWMwOTkzOTg2MzMxMTQ0ZTk
From: <sip:julien@voip.cahri.net>;tag=e48f0c01
To: <sip:0170809700@voip.cahri.net>;tag=bc3a3d1d-a2eb-4338-aa6f-52e9dc3e9ac7
CSeq: 3 BYE
Server: Asterisk PBX 14.1.2
Content-Length:  0


    -- Channel PJSIP/julien-00000006 left 'simple_bridge' basic-bridge <f292e1f7-f316-46f6-94ae-c838f0360bfc>
    -- Channel IAX2/plugandtel-15756 left 'simple_bridge' basic-bridge <f292e1f7-f316-46f6-94ae-c838f0360bfc>
  == Spawn extension (macro-dialout, s, 1) exited non-zero on 'PJSIP/julien-00000006' in macro 'dialout'
    -- Hungup 'IAX2/plugandtel-15756'
  == Spawn extension (dial-sip, 0170809700, 2) exited non-zero on 'PJSIP/julien-00000006'
<--- Received SIP request (771 bytes) from UDP:197.225.152.105:61429 --->
BYE sip:41.213.137.68:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:61429;branch=z9hG4bK-524287-1---46169407acf8c62a;rport;alias
Max-Forwards: 70
Contact: <sip:julien@197.225.152.105:61429;rinstance=8c06fb3fbe8c1a2c>
To: <sip:0170809700@voip.cahri.net>;tag=bc3a3d1d-a2eb-4338-aa6f-52e9dc3e9ac7
From: <sip:julien@voip.cahri.net>;tag=e48f0c01
Call-ID: ZThhNGI0MTk4YmY4MWQ3OWMwOTkzOTg2MzMxMTQ0ZTk
CSeq: 3 BYE
User-Agent: Bria iOS release 3.8.1 stamp 36276.36277
Authorization: Digest username="julien",realm="asterisk",nonce="1479226444/9439e99a804765a02cb705211c16abe9",uri="sip:41.213.137.68:5060",response="a369b6a39720aec236e3e08b65d53566",cnonce="9e2016bc8386af5f437e726a66978d74",nc=00000002,qop=auth,algorithm=md5,opaque="06a6ba4912dab6ac"
Content-Length: 0


<--- Transmitting SIP response (382 bytes) to UDP:197.225.152.105:61429 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:61429;rport=61429;received=197.225.152.105;branch=z9hG4bK-524287-1---46169407acf8c62a;alias
Call-ID: ZThhNGI0MTk4YmY4MWQ3OWMwOTkzOTg2MzMxMTQ0ZTk
From: <sip:julien@voip.cahri.net>;tag=e48f0c01
To: <sip:0170809700@voip.cahri.net>;tag=bc3a3d1d-a2eb-4338-aa6f-52e9dc3e9ac7
CSeq: 3 BYE
Server: Asterisk PBX 14.1.2
Content-Length:  0


voip*CLI> 

Thanks in advance for your insights :slight_smile:

Capture your RTP traffic then try to analyze it in Wireshark. Most likely you will notice different Opus ‘flavors’ on transmit and receive.