Trunk SIP - Problème One way audio

**ddpetit** · 08/08/2019, 13h26

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

Discussion: Trunk SIP - Problème One way audio

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