Trunk SIP - Problème One way audio

[ddpetit]

<------------>
Audio is at 10246
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 91.121.129.23:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-GJGF-8707921a-3fcd69fb;received=91.121.129.23;rport=5060
Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
CSeq: 198813421 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:s@185.246.18.202:5060>
Content-Type: application/sdp
Content-Length: 268

v=0
o=root 665561663 665561665 IN IP4 185.246.18.202
s=Asterisk PBX 13.14.1~dfsg-2+deb9u4
c=IN IP4 185.246.18.202
t=0 0
m=audio 10246 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

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

<--- SIP read from UDP:91.121.129.23:5060 --->
ACK sip:s@192.168.2.100:5060 SIP/2.0
Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
Contact: <sip:10.7.1.65:5060>
CSeq: 198813421 ACK
From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
Max-Forwards: 29
To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-SAQW-8707921f-562ea85e
User-Agent: Cirpack/v4.76 (gw_sip)
Content-Length: 0

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

<--- SIP read from UDP:192.168.130.49:5060 --->
BYE sip:0383723596@192.168.130.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.130.49:5060;branch=z9hG4bK144884565
From: <sip:214@192.168.130.49:5060>;tag=2669272485
To: "+33383723596" <sip:0383723596@192.168.130.254>;tag=as3efc2963
Call-ID: 56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060
CSeq: 2 BYE
Contact: <sip:214@192.168.130.49:5060>
Max-Forwards: 70
User-Agent: Yealink SIP-T41S 66.84.0.15
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.130.49:5060 (NAT)
Scheduling destruction of SIP dialog '56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060' in 6400 ms (Method: BYE)

<--- Transmitting (NAT) to 192.168.130.49:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.130.49:5060;branch=z9hG4bK144884565;received=192.168.130.49;rport=5060
From: <sip:214@192.168.130.49:5060>;tag=2669272485
To: "+33383723596" <sip:0383723596@192.168.130.254>;tag=as3efc2963
Call-ID: 56f568e446b3e03a1ee90e065323974a@192.168.130.254:5060
CSeq: 2 BYE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '19122-IS-0c93dd04-597706255@siptrunk.ovh.net' in 6400 ms (Method: ACK)
Reliably Transmitting (NAT) to 91.121.129.23:5060:
BYE sip:10.7.1.65:5060 SIP/2.0
Via: SIP/2.0/UDP 185.246.18.202:5060;branch=z9hG4bK26126eb9;rport
Route: <sip:91.121.129.23:5060;lr>
Max-Forwards: 70
From: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
To: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
CSeq: 102 BYE
User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u4
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:91.121.129.23:5060 --->
SIP/2.0 200 OK
Call-ID: 19122-IS-0c93dd04-597706255@siptrunk.ovh.net
CSeq: 102 BYE
From: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
Record-Route: <sip:91.121.129.23:5060;transport=udp;lr>;session=443934
To: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
Via: SIP/2.0/UDP 192.168.2.100:5060;received=192.168.2.100;rport=5060;branch=z9hG4bK26126eb9
Server: Cirpack/v4.76 (gw_sip)
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '19122-IS-0c93dd04-597706255@siptrunk.ovh.net' Method: ACK

[bwen]

Bonjour,

J'ai déjà rencontré ce problème plusieurs, c'est généralement toujours un problème "réseau" dans 99% des cas, une fois c’était effectivement un blocage niveau firewall et l'autre cas que j'ai rencontré était un peu plus "tordu", c’était une route de retour qui n'empruntait pas le même chemin que la requête initiale.

Cordialement,