one way audio or no audio

[davide.filippi]

Hi all, sometime, in my call center there are audio issues such as one way audio or no audio and when it's appen, i see no rtp traffic.

this is the sip debug and the rtp debug (that not exist).


<--- SIP read from UDP:192.168.1.181:39528 --->
INVITE sip:393272421889@192.168.100.221:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.181:39528;branch=z9hG4bK-d87543-793fbd589a1daa4d-1--d 87543-;rport
Max-Forwards: 70
Contact: <sip:VOIPTEST@192.168.1.181:39528>
To: "393272421889"<sip:393272421889@192.168.100.221:5060>
From: "VOIPTEST"<sip:VOIPTEST@192.168.100.221:5060>;tag=8d2c617e
Call-ID: MDIyNzY4MWVmMGYzZTU5MmRiMDA2OTZlOTJiMTFhMzc.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INF O
Content-Type: application/sdp
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 473

v=0
o=- 2 2 IN IP4 192.168.1.181
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.181
t=0 0
m=audio 44448 RTP/AVP 107 119 100 106 0 105 98 8 101
a=alt:1 2 : eQpofyBA 4H3455IQ 172.31.80.21 44448
a=alt:2 1 : y5nHv7II FaPO0ir2 192.168.1.181 44448
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (12 headers 17 lines) ---
Sending to 192.168.1.181:39528 (no NAT)
Sending to 192.168.1.181:39528 (no NAT)
Using INVITE request as basis request - MDIyNzY4MWVmMGYzZTU5MmRiMDA2OTZlOTJiMTFh Mzc.
Found peer 'VOIPTEST' for 'VOIPTEST' from 192.168.1.181:39528
== Using SIP RTP CoS mark 5
Found RTP audio format 107
Found RTP audio format 119
Found RTP audio format 100
Found RTP audio format 106
Found RTP audio format 0
Found RTP audio format 105
Found RTP audio format 98
Found RTP audio format 8
Found RTP audio format 101
Found unknown media description format BV32 for ID 107
Found unknown media description format BV32-FEC for ID 119
Found audio description format SPEEX for ID 100
Found unknown media description format SPEEX-FEC for ID 106
Found unknown media description format SPEEX-FEC for ID 105
Found audio description format iLBC for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw|speex16|ilbc)/video= (nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephon e-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.181:44448
Looking for 393272421889 in billing (domain 192.168.100.221)
list_route: hop: <sip:VOIPTEST@192.168.1.181:39528>

<--- Transmitting (NAT) to 192.168.1.181:39528 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.181:39528;branch=z9hG4bK-d87543-793fbd589a1daa4d-1--d 87543-;received=192.168.1.181;rport=39528
From: "VOIPTEST"<sip:VOIPTEST@192.168.100.221:5060>;tag=8d2c617e
To: "393272421889"<sip:393272421889@192.168.100.221:5060>
Call-ID: MDIyNzY4MWVmMGYzZTU5MmRiMDA2OTZlOTJiMTFhMzc.
CSeq: 1 INVITE
Server: Asterisk PBX 11.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
Supported: replaces, timer
Contact: <sip:393272421889@192.168.100.221:5060>
Content-Length: 0


<------------>
-- Executing [393272421889@billing:1] AGI("SIP/VOIPTEST-000000b8", ""/var/ww w/html/mbilling/agi.php"") in new stack
-- Launched AGI Script /var/www/html/mbilling/agi.php
== Manager 'magnus' logged on from 127.0.0.1
== Manager 'magnus' logged off from 127.0.0.1
-- AGI Script Executing Application: (DIAL) Options: (sip/GO_VoIP_1/00393272 421889,60,L(2147483647:61000:30000))
== Using SIP RTP CoS mark 5
-- Called sip/GO_VoIP_1/00393272421889
-- SIP/GO_VoIP_1-000000b9 is making progress passing it to SIP/VOIPTEST-0000 00b8
Audio is at 33636
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (NAT) to 192.168.1.181:39528 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.181:39528;branch=z9hG4bK-d87543-793fbd589a1daa4d-1--d 87543-;received=192.168.1.181;rport=39528
From: "VOIPTEST"<sip:VOIPTEST@192.168.100.221:5060>;tag=8d2c617e
To: "393272421889"<sip:393272421889@192.168.100.221:5060>;tag=as3da93142
Call-ID: MDIyNzY4MWVmMGYzZTU5MmRiMDA2OTZlOTJiMTFhMzc.
CSeq: 1 INVITE
Server: Asterisk PBX 11.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS H, MESSAGE
Supported: replaces, timer
Contact: <sip:393272421889@192.168.100.221:5060>
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 565957231 565957231 IN IP4 192.168.100.221
s=Asterisk PBX 11.25.0
c=IN IP4 192.168.100.221
t=0 0
m=audio 33636 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
== Manager 'magnus' logged on from 127.0.0.1
== Manager 'magnus' logged off from 127.0.0.1
== Manager 'magnus' logged on from 127.0.0.1
== Manager 'magnus' logged off from 127.0.0.1
-- SIP/GO_VoIP_1-000000b9 is ringing

<--- Transmitting (NAT) to 192.168.1.181:39528 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.181:39528;branch=z9hG4bK-d87543-793fbd589a1daa4d-1--d87543-;received=192.168.1.181;rport=39528
From: "VOIPTEST"<sip:VOIPTEST@192.168.100.221:5060>;tag=8d2c617e
To: "393272421889"<sip:393272421889@192.168.100.221:5060>;tag=as3da93142
Call-ID: MDIyNzY4MWVmMGYzZTU5MmRiMDA2OTZlOTJiMTFhMzc.
CSeq: 1 INVITE
Server: Asterisk PBX 11.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:393272421889@192.168.100.221:5060>
Content-Length: 0


<------------>
-- SIP/GO_VoIP_1-000000b9 is making progress passing it to SIP/VOIPTEST-000000b8
== Manager 'magnus' logged on from 127.0.0.1
== Manager 'magnus' logged off from 127.0.0.1
-- SIP/GO_VoIP_1-000000b9 answered SIP/VOIPTEST-000000b8
Audio is at 33636
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 192.168.1.181:39528 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.181:39528;branch=z9hG4bK-d87543-793fbd589a1daa4d-1--d87543-;received=192.168.1.181;rport=39528
From: "VOIPTEST"<sip:VOIPTEST@192.168.100.221:5060>;tag=8d2c617e
To: "393272421889"<sip:393272421889@192.168.100.221:5060>;tag=as3da93142
Call-ID: MDIyNzY4MWVmMGYzZTU5MmRiMDA2OTZlOTJiMTFhMzc.
CSeq: 1 INVITE
Server: Asterisk PBX 11.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:393272421889@192.168.100.221:5060>
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 565957231 565957231 IN IP4 192.168.100.221
s=Asterisk PBX 11.25.0
c=IN IP4 192.168.100.221
t=0 0
m=audio 33636 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
-- Locally bridging SIP/VOIPTEST-000000b8 and SIP/GO_VoIP_1-000000b9

<--- SIP read from UDP:192.168.1.181:39528 --->
ACK sip:393272421889@192.168.100.221:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.181:39528;branch=z9hG4bK-d87543-83341c1f68465833-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:VOIPTEST@192.168.1.181:39528>
To: "393272421889"<sip:393272421889@192.168.100.221:5060>;tag=as3da93142
From: "VOIPTEST"<sip:VOIPTEST@192.168.100.221:5060>;tag=8d2c617e
Call-ID: MDIyNzY4MWVmMGYzZTU5MmRiMDA2OTZlOTJiMTFhMzc.
CSeq: 1 ACK
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 0


please help