[asterisk-users] Getting one way audio even NAT is configured
Ahmed Munir
ahmedmunir007 at gmail.com
Wed Feb 1 13:16:45 CST 2012
Hi all,
I'm getting one way audio when calling over the SIP trunk i.e. end device B
(remote end of SIP trunk) can hear device A (softphone registered with
Asterisk) but device A can't hear device B. Even though I configured same
NAT configurations on other servers and they are working good. The NAT
configuration is listed below;
localnet=130.0.0.0/130.0.0.0
externhost=12.131.12.13
externrefresh=10
fromdomain=test.localhost.com
nat=yes
qualify=yes
canreinvite=no
NAT on device end i.e. my softphone (extension) has already set to yes with
canreinvite=no but still unable to resolve this issue. SIP traces are
listed below;
Reliably Transmitting (NAT) to 12.194.12.12:5060:
INVITE sip:173242 at 12.194.12.12 SIP/2.0
Via: SIP/2.0/UDP 12.131.12.13:5060;branch=z9hG4bK1fbbab95;rport
Max-Forwards: 70
From: "77057" <sip:77057 at test.localhost.com>;tag=as1fa9b502
To: <sip:173242 at 12.194.12.12>
Contact: <sip:77057 at 12.131.12.13:5060>
Call-ID: 04ce1d566f1f17a221caba261e2af4bb at test.localhost.com
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.8.5.0)
Date: Wed, 01 Feb 2012 16:11:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 285
v=0
o=root 122642112 122642112 IN IP4 12.131.12.13
s=Asterisk PBX 1.8.5.0
c=IN IP4 12.131.12.13
t=0 0
m=audio 16006 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- Called SIP/ATTLABS-IP-FlexReach/173242
<--- SIP read from UDP:12.194.12.12:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 12.131.12.13:5060
;received=12.131.12.13;branch=z9hG4bK1fbbab95;rport=5060
From: "77057" <sip:77057 at test.localhost.com>;tag=as1fa9b502
To: <sip:173242 at 12.194.12.12>
Call-ID: 04ce1d566f1f17a221caba261e2af4bb at test.localhost.com
CSeq: 102 INVITE
<------------->
--- (6 headers 0 lines) ---
<--- SIP read from UDP:12.194.12.12:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 12.131.12.13:5060
;received=12.131.12.13;branch=z9hG4bK1fbbab95;rport=5060
From: "77057" <sip:77057 at test.localhost.com>;tag=as1fa9b502
To: <sip:173242 at 12.194.12.12
>;tag=SDk39gc99-7026517720142726_c2b08.2.2.1323416803184.0_1194082_2371400
Call-ID: 04ce1d566f1f17a221caba261e2af4bb at test.localhost.com
CSeq: 102 INVITE
Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK
Contact: <sip:12.194.12.12:5060;transport=udp>
Content-Length: 237
Content-Disposition: session; handling=required
Content-Type: application/sdp
v=0
o=Sonus_UAC 14862 4757 IN IP4 12.194.12.12
s=SIP Media Capabilities
c=IN IP4 12.194.12.12
t=0 0
m=audio 16534 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
<------------->
--- (11 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0
(nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 12.194.12.12:16534
-- SIP/ATTLABS-IP-FlexReach-00000025 is making progress passing it to
SIP/2005-00000024
<--- SIP read from UDP:12.194.12.12:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 12.131.12.13:5060
;received=12.131.12.13;branch=z9hG4bK1fbbab95;rport=5060
From: "77057" <sip:77057 at test.localhost.com>;tag=as1fa9b502
To: <sip:173242 at 12.194.12.12
>;tag=SDk39gc99-7026517720142726_c2b08.2.2.1323416803184.0_1194082_2371400
Call-ID: 04ce1d566f1f17a221caba261e2af4bb at test.localhost.com
CSeq: 102 INVITE
Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK
Accept: application/sdp, application/isup, application/dtmf,
application/dtmf-relay, multipart/mixed
Contact: <sip:12.194.12.12:5060;transport=udp>
Content-Length: 237
Content-Disposition: session; handling=required
Content-Type: application/sdp
v=0
o=Sonus_UAC 14862 4757 IN IP4 12.194.12.12
s=SIP Media Capabilities
c=IN IP4 12.194.12.12
t=0 0
m=audio 16534 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
<------------->
--- (12 headers 11 lines) ---
list_route: hop: <sip:12.194.12.12:5060;transport=udp>
set_destination: Parsing <sip:12.194.12.12:5060;transport=udp> for
address/port to send to
set_destination: set destination to 12.194.12.12:5060
Transmitting (NAT) to 12.194.12.12:5060:
ACK sip:12.194.12.12:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 12.131.12.13:5060;branch=z9hG4bK483f052d;rport
Max-Forwards: 70
From: "77057" <sip:77057 at test.localhost.com>;tag=as1fa9b502
To: <sip:173242 at 12.194.12.12
>;tag=SDk39gc99-7026517720142726_c2b08.2.2.1323416803184.0_1194082_2371400
Contact: <sip:77057 at 12.131.12.13:5060>
Call-ID: 04ce1d566f1f17a221caba261e2af4bb at test.localhost.com
CSeq: 102 ACK
User-Agent: FPBX-2.9.0(1.8.5.0)
Content-Length: 0
---
-- SIP/ATTLABS-IP-FlexReach-00000025 answered SIP/2005-00000024
-- Locally bridging SIP/2005-00000024 and
SIP/ATTLABS-IP-FlexReach-00000025
Reliably Transmitting (NAT) to 12.194.12.12:5060:
OPTIONS sip:12.194.12.12 SIP/2.0
Via: SIP/2.0/UDP 12.131.12.13:5060;branch=z9hG4bK06532068;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown at test.localhost.com>;tag=as054a7d2d
To: <sip:12.194.12.12>
Contact: <sip:Unknown at 12.131.12.13:5060>
Call-ID: 767dcb7d4406d06c248a7056559ad301 at test.localhost.com
CSeq: 102 OPTIONS
User-Agent: FPBX-2.9.0(1.8.5.0)
Date: Wed, 01 Feb 2012 16:11:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:12.194.12.12:5060 --->
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 12.131.12.13:5060
;received=12.131.12.13;branch=z9hG4bK06532068;rport=5060
From: "Unknown" <sip:Unknown at test.localhost.com>;tag=as054a7d2d
To: <sip:12.194.12.12>;tag=aprqngfrt-d1v40r10000c6
Call-ID: 767dcb7d4406d06c248a7056559ad301 at test.localhost.com
CSeq: 102 OPTIONS
Reason: Q.850;cause=55;text="Call Terminated"
Allow: INVITE,ACK,BYE,CANCEL,PRACK,INFO,REFER,UPDATE,MESSAGE,PUBLISH
<------------->
--- (8 headers 0 lines) ---
Scheduling destruction of SIP dialog '
04ce1d566f1f17a221caba261e2af4bb at test.localhost.com' in 6400 ms (Method:
INVITE)
set_destination: Parsing <sip:12.194.12.12:5060;transport=udp> for
address/port to send to
set_destination: set destination to 12.194.12.12:5060
Reliably Transmitting (NAT) to 12.194.12.12:5060:
BYE sip:12.194.12.12:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 12.131.12.13:5060;branch=z9hG4bK2ab85b31;rport
Max-Forwards: 70
From: "77057" <sip:77057 at test.localhost.com>;tag=as1fa9b502
To: <sip:173242 at 12.194.12.12
>;tag=SDk39gc99-7026517720142726_c2b08.2.2.1323416803184.0_1194082_2371400
Call-ID: 04ce1d566f1f17a221caba261e2af4bb at test.localhost.com
CSeq: 103 BYE
User-Agent: FPBX-2.9.0(1.8.5.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:12.194.12.12:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 12.131.12.13:5060
;received=12.131.12.13;branch=z9hG4bK2ab85b31;rport=5060
From: "77057" <sip:77057 at test.localhost.com>;tag=as1fa9b502
To: <sip:173242 at 12.194.12.12
>;tag=SDk39gc99-7026517720142726_c2b08.2.2.1323416803184.0_1194082_2371400
Call-ID: 04ce1d566f1f17a221caba261e2af4bb at test.localhost.com
CSeq: 103 BYE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '
04ce1d566f1f17a221caba261e2af4bb at test.localhost.com' Method: INVITE
The Asterisk version I'm using is 1.8.5. Please assist me at earliest.
--
Regards,
Ahmed Munir Chohan
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