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Ahmed Munir

2012-Feb-01 19:16 UTC

[asterisk-users] Getting one way audio even NAT is configured

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_2371400Call-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_2371400Call-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_2371400Contact: <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_2371400Call-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_2371400Call-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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Warren Selby

2012-Feb-01 20:38 UTC

head link

[asterisk-users] Getting one way audio even NAT is configured

On Wed, Feb 1, 2012 at 1:16 PM, Ahmed Munir <ahmedmunir007 at gmail.com>
wrote:
> 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;
>
><snip>

>
> The Asterisk version I'm using is 1.8.5. Please assist me at earliest.
>
Which device (A or B) is behind NAT with regards to your asterisk server?
Is that the actual localnet= statement you're using, because to my
understanding that is not the proper format to use (should be
localnet=x.x.x.x/y.y.y.y where x.x.x.x is your actual local network, and
y.y.y.y is your subnet for your local network).

-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com <http://www.selbytech.com>
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Ahmed Munir

2012-Feb-02 16:59 UTC

head link

[asterisk-users] Getting one way audio even NAT is configured

Hi Warren,

Device A is behind NAT with regards to asterisk server. As far as localnet
statement first I did configured localnet = 130.8.2.0/255.255.255.0 as per
local network, after that made a SIP call and the message I'm getting is
listed below;

[Feb  2 11:14:52] WARNING[23868]: chan_sip.c:3622 retrans_pkt:
Retransmission timeout reached on transmission
OGEzODA0MzI4MWE1NzdiZDlkNmQ3NjYyMzJjYzYyOTY. for seqno 1 (Critical
Response) -- See
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[Feb  2 11:14:52] WARNING[23868]: chan_sip.c:3651 retrans_pkt: Hanging up
call OGEzODA0MzI4MWE1NzdiZDlkNmQ3NjYyMzJjYzYyOTY. - no reply to our
critical packet (see
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

So after setting to 130.0.0.0/130.0.0.0 I wasn't getting the above warning
message but facing one way audio.

Date: Wed, 1 Feb 2012 14:38:01 -0600> From: Warren Selby <wcselby at selbytech.com>
> Subject: Re: [asterisk-users] Getting one way audio even NAT is
>        configured
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>        <asterisk-users at lists.digium.com>
> Message-ID:
>        <CAM_w8OJmX0nXfdU06p=-fpRAbZ2h7Tqr-mJMJNFnWeAvkjS+ZA at
mail.gmail.com
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> On Wed, Feb 1, 2012 at 1:16 PM, Ahmed Munir <ahmedmunir007 at
gmail.com>
> wrote:
>
> > 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;
> >
> >
> <snip>
>
>
> >
> > The Asterisk version I'm using is 1.8.5. Please assist me at
earliest.
> >
>
> Which device (A or B) is behind NAT with regards to your asterisk server?
> Is that the actual localnet= statement you're using, because to my
> understanding that is not the proper format to use (should be
> localnet=x.x.x.x/y.y.y.y where x.x.x.x is your actual local network, and
> y.y.y.y is your subnet for your local network).
>
> --
> Thanks,
> --Warren Selby, dCAP
> http://www.SelbyTech.com <http://www.selbytech.com>
>
--
Regards,

Ahmed Munir Chohan
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