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Discussion: Trunk SIP - Problème One way audio

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  1. 08/08/2019, 13h26 #1
    ddpetit
    ddpetit est déconnecté
    Membre Junior
    Date d'inscription
    juillet 2012
    Messages
    14
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    Code:
    ---
Audio is at 10246
Adding codec ulaw to SDP
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-TKLX-87078c4d-1549530f;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: 198813419 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: 292

v=0
o=root 665561663 665561663 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 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

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

<--- 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: 198813419 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-LHLE-8707918f-3db175f4
User-Agent: Cirpack/v4.76 (gw_sip)
Content-Length: 0

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

<--- SIP read from UDP:91.121.129.23:5060 --->
INVITE 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>
Content-Type: application/sdp
CSeq: 198813420 INVITE
From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
Max-Forwards: 29
Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-BPIK-87079191-6277c4a6
Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
User-Agent: Cirpack/v4.76 (gw_sip)
Content-Length: 318

v=0
o=cp10 156526349186 156526349187 IN IP4 91.121.128.136
s=SIP Call
c=IN IP4 91.121.128.136
t=0 0
m=audio 32142 RTP/AVP 0 8 18 101
b=AS:82
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
--- (13 headers 15 lines) ---
Sending to 91.121.129.23:5060 (NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 91.121.128.136:32142

<--- Transmitting (NAT) to 91.121.129.23:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-BPIK-87079191-6277c4a6;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: 198813420 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-Length: 0


<------------>
Audio is at 10246
Adding codec ulaw to SDP
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-BPIK-87079191-6277c4a6;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: 198813420 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: 292

v=0
o=root 665561663 665561664 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 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

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

<--- 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: 198813420 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-YYHU-87079197-2495cec1
User-Agent: Cirpack/v4.76 (gw_sip)
Content-Length: 0

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

<--- SIP read from UDP:91.121.129.23:5060 --->
INVITE 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>
Content-Type: application/sdp
CSeq: 198813421 INVITE
From: "+33383723596" <sip:0383723596@siptrunk.ovh.net;user=phone>;tag=19122-KP-0c93dd05-7cea4df27
Max-Forwards: 29
Record-Route: <sip:91.121.129.23:5060;lr>;session=443934
To: <sip:0383720044@10.7.1.65;user=phone>;tag=as613b4868
Via: SIP/2.0/UDP 91.121.129.23:5060;branch=z9hG4bK-GJGF-8707921a-3fcd69fb
Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
User-Agent: Cirpack/v4.76 (gw_sip)
Content-Length: 243

v=0
o=cp10 156526349186 156526349188 IN IP4 91.121.128.136
s=SIP Call
c=IN IP4 91.121.128.136
t=0 0
m=audio 32142 RTP/AVP 8 101
b=AS:82
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
--- (13 headers 12 lines) ---
Sending to 91.121.129.23:5060 (NAT)
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 91.121.128.136:32142

<--- Transmitting (NAT) to 91.121.129.23:5060 --->
SIP/2.0 100 Trying
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-Length: 0
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  2. 08/08/2019, 13h27 #2
    ddpetit
    ddpetit est déconnecté
    Membre Junior
    Date d'inscription
    juillet 2012
    Messages
    14
    Downloads
    0
    Uploads
    0
    Code:
    <------------>
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
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  3. 09/08/2019, 16h55 #3
    bwen
    bwen est déconnecté
    Membre Junior
    Date d'inscription
    août 2016
    Messages
    13
    Downloads
    0
    Uploads
    0
    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,
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