One way audio (Asterisk v13 bug)
Asterisk server wrongly select audio codec which causes one-way audio issue. It could be clearly seen in the SIP log (please see example below). In the example below, instead of PCMU (0) which was negotiated, Asterisk is sending packets in PCMA (8). We believe this is connected to the bug in Asterisk version 13 which was mentioned on the forums, for example:Â Asterisk 13.13 Codec selection : Possible Bug
To mitigate this, we recommend disabling all the codes in Ringotel connection settings except PCMU.
Alternatively, you can try setting identical codecs order in Ringotel connection and PBX settings.
Example:
------ Outgoing call to sip:01510000000@voip.domain.net:5061;transport=tls -------------------------
17/06 XXX.X.XXX.XXX
INVITE sip:01510000000@voip.domain.net:5061;transport=tls SIP/2.0
Call-ID: d7762e90bfa148e37ea41f4bd4075a
Via: SIP/2.0/TLS XXX.X.XXX.XXX:5061;branch=z9hG4bKf322913afb602e71
From: "Jim Smith" <sip:jim@voip.domain.net:5061>;tag=6588be27
To: "01510000000" <sip:01510000000@voip.domain.net:5061;transport=tls>
CSeq: 2061 INVITE
Max-Forwards: 70
User-Agent: Ringotel Server
Supported: replaces
Contact: <sip:jim@XXX.X.XXX.XXX:5061;transport=tls>
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, UPDATE, REFER, INFO, SUBSCRIBE, NOTIFY, MESSAGE
Content-Type: application/sdp
Content-Length: 248
v=0
o=jim 1655472689 1 IN IP4 XXX.X.XXX.XXX
s=SIP Call
c=IN IP4 XXX.X.XXX.XXX
t=0 0
m=audio 43892 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
…
17/06 22.22.22.222Â
SIP/2.0 183 Session Progress
Call-ID: d7762e90bfa148e37ea41f4bd4075a
Via: SIP/2.0/TLS XXX.X.XXX.XXX:5061;rport=37421;branch=z9hG4bKf96003e8c181348f
From: "Jim Smith" <sip:jim@voip.domain.net>;tag=6588be27
To: "01510000000" <sip:01510000000@voip.domain.net>;tag=d3930f95-ad02-43bd-868a-833b4d4dc29f
CSeq: 2062 INVITE
Server: Asterisk PBX 13.4.0
Record-Route: <sip:127.0.0.8;line=sr-N6IAzBF4zmZ6OBFLWGZfMxNXoUVXNBVQo6ZXgSIqHRaJWGZZz.qfz6srpxusg9M*>, <sip:XXX.X.XXX.XXX:5061;r2=on;nat=yes;lr;transport=tls;did=db4.8991>
Contact: <sip:127.0.0.8;line=sr-N6IAzBF4zmZ6OBFLMJZfzxy-W.y6My**>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER
Content-Type: application/sdp
Content-Length: 272
v=0
o=- 1655472689 3 IN IP4 XXX.X.XXX.XXX
s=Asterisk
c=IN IP4 XXX.X.XXX.XXX
t=0 0
m=audio 59068 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=nortpproxy:yes
...
17/06 XX.XX.XX.XXX
ACK sip:127.0.0.8;line=sr-N6IAzBF4zmZ6OBFLMJZfzxy-W.y6My**;transport=tls SIP/2.0
Call-ID: d7762e90bfa148e37ea41f4bd4075a
Via: SIP/2.0/TLS XX.XX.XX.XXX:5061;branch=z9hG4bK90f1854c843811c4
From: "Jim Smith" <sip:jim@voip.domain.net:5061>;tag=6588be27
To: "01510000000" <sip:01510000000@voip.domain.net:5061;transport=tls>;tag=d3930f95-ad02-43bd-868a-833b4d4dc29f
CSeq: 2062 ACK
Max-Forwards: 70
User-Agent: Ringotel Server
Proxy-Authorization: Digest response="df29d4cc5debf71a7ec415539e8c47ae",realm="voip.domain.net",uri="sip:01510000000@voip.domain.net:5061;transport=tls",nonce="YqyDXWKsgjFwjIDbyX94FSYjTzc3I774",username="Jim Smith"
Route: <sip:XXX.X.XXX.XXX:5061;r2=on;nat=yes;lr;transport=tls;did=db4.8991>, <sip:127.0.0.8;line=sr-N6IAzBF4zmZ6OBFLWGZfMxNXoUVXNBVQo6ZXgSIqHRaJWGZZz.qfz6srpxusg9M*>
Contact: <sip:Jim Smith@XX.XX.XXX.XXX:5061;transport=tls>
Content-Type: application/sdp
Content-Length: 224
v=0
o=jim 1655472689 2 IN IP4 XX.XXX.XX.XXX
s=SIP Call
c=IN IP4 XX.XXX.XX.XXX
t=0 0
m=audio 43892 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
…
17/06 XX.XX.XX.XXX Session details:
ssrc=375200669 frac=0 psize=160 fseq=19645 delay=0 jit=10 packets=540 mfrac=0 codec=0 lseq=20184 octets=86400 mdelay=0 lost=0 To=XXX.X.XXX.XXX:59068Â
ssrc=1768293115 mjit=71 frac=0 From=XXX.X.XXX.XXX:59068 psize=160 fseq=40182 jit=15 packets=620 mfrac=0 codec=8 lseq=40801 octets=99200 mdelay=13 lost=0Â
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