OK, here's a trace from the calling handset. I see an Unauthorized, but
presume that is simply part of the authentication negotiation. However, you
can also see the 3.5s delay between the replies "Trying" and "Ringing". What
is causing that? Will try to get a trace at the Astlinux end, and the
receiving handset to compare.
Thanks
Tom
Sent to udp:10.8.243.5:5060 at 19/8/2013 17:37:41:906 (1143 bytes):
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.8.242.30:2048;branch=z9hG4bK-8j6smi3er0dr;rport
From: "611530" <sip:[email protected]>;tag=tg9uljgf9b
To: <sip:[email protected];user=phone>
Call-ID: 521249d39631-xegxs8xyranx
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:[email protected]:2048;line=0xts5ibu>;reg-id=1
X-Serialnumber: 000413255954
P-Key-Flags: keys="3"
User-Agent: snom300/8.7.3.19
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 401
v=0
o=root 1482026541 1482026541 IN IP4 10.8.242.30
s=call
c=IN IP4 10.8.242.30
t=0 0
m=audio 64486 RTP/AVP 9 0 8 3 99 108 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:108 AAL2-G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
_____
Received from udp:10.8.243.5:5060 at 19/8/2013 17:37:42:015 (519 bytes):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
10.8.242.30:2048;branch=z9hG4bK-8j6smi3er0dr;received=10.8.242.30;rport=2048
From: "611530" <sip:[email protected]>;tag=tg9uljgf9b
To: <sip:[email protected];user=phone>;tag=as2d7cd3c8
Call-ID: 521249d39631-xegxs8xyranx
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.22.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0ab35520"
Content-Length: 0
_____
Sent to udp:10.8.243.5:5060 at 19/8/2013 17:37:42:027 (370 bytes):
ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.8.242.30:2048;branch=z9hG4bK-8j6smi3er0dr;rport
From: "611530" <sip:[email protected]>;tag=tg9uljgf9b
To: <sip:[email protected];user=phone>;tag=as2d7cd3c8
Call-ID: 521249d39631-xegxs8xyranx
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:[email protected]:2048;line=0xts5ibu>;reg-id=1
Content-Length: 0
_____
Sent to udp:10.8.243.5:5060 at 19/8/2013 17:37:42:065 (1312 bytes):
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 10.8.242.30:2048;branch=z9hG4bK-zov9bsin9wgw;rport
From: "611530" <sip:[email protected]>;tag=tg9uljgf9b
To: <sip:[email protected];user=phone>
Call-ID: 521249d39631-xegxs8xyranx
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <sip:[email protected]:2048;line=0xts5ibu>;reg-id=1
X-Serialnumber: 000413255954
P-Key-Flags: keys="3"
User-Agent: snom300/8.7.3.19
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Authorization: Digest
username="611530",realm="asterisk",nonce="0ab35520",uri="sip:[email protected];
user=phone",response="c4c6d813402c4346063ff1da0f521583",algorithm=MD5
Content-Type: application/sdp
Content-Length: 401
v=0
o=root 1482026541 1482026541 IN IP4 10.8.242.30
s=call
c=IN IP4 10.8.242.30
t=0 0
m=audio 64486 RTP/AVP 9 0 8 3 99 108 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:99 G726-32/8000
a=rtpmap:108 AAL2-G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
_____
Received from udp:10.8.243.5:5060 at 19/8/2013 17:37:42:084 (495 bytes):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.8.242.30:2048;branch=z9hG4bK-zov9bsin9wgw;received=10.8.242.30;rport=2048
From: "611530" <sip:[email protected]>;tag=tg9uljgf9b
To: <sip:[email protected];user=phone>
Call-ID: 521249d39631-xegxs8xyranx
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.22.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Length: 0
_____
Received from udp:10.8.243.5:5060 at 19/8/2013 17:37:45:589 (511 bytes):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
10.8.242.30:2048;branch=z9hG4bK-zov9bsin9wgw;received=10.8.242.30;rport=2048
From: "611530" <sip:[email protected]>;tag=tg9uljgf9b
To: <sip:[email protected];user=phone>;tag=as21ede872
Call-ID: 521249d39631-xegxs8xyranx
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.22.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Length: 0
From: The Cadillac Kid [mailto:[email protected]]
Sent: 19 August 2013 17:28
To: AstLinux Users Mailing List
Subject: Re: [Astlinux-users] Delay before audio is audible
tcpdump is great if the packets are actually making it to the astlinux box..
but if they are getting lost somewhere in the network you will not see
them...
-Christopher
_____
From: Tom Chadwin <[email protected]>
To: 'AstLinux Users Mailing List' <[email protected]>
Sent: Monday, August 19, 2013 11:54 AM
Subject: Re: [Astlinux-users] Delay before audio is audible
No packet loss that I can see. Looking into how to get a pcap SIP trace as
Kris asked. What's best given this is just an up-to-date Astlinux 1.8
install? tcpdump?
Thanks
Tom
> -----Original Message-----
> From: The Cadillac Kid [mailto:[email protected]]
> Sent: 19 August 2013 16:20
> To: [email protected]
> Subject: Re: [Astlinux-users] Delay before audio is audible
>
>
>
>
> If you are using direct media = no then you may get some insight with sip
> show channelstats in the asterisk cli to see if you are losing RTP
packets..
>
> Christopher
>
> ------------------------------
> On Mon, Aug 19, 2013 9:57 AM EDT Kristian Kielhofner wrote:
>
> >Tom,
> >
> > Are you able to provide a pcap with a SIP trace?
> >
> >On Mon, Aug 19, 2013 at 8:37 AM, Tom Chadwin
> ><[email protected]> wrote:
> >> Hello again
> >>
> >> I set directmedia=no, and this seemed to fix the issue. However, it has
> now
> >> reoccurred. What else could cause this? Silence (no ringtone for
caller) for
> >> some seconds, then audio establishes. Could it be something to do with
> our
> >> phones (Snom 300)? I had to disable SRTP (I think it was)to get them to
> work
> >> at all when I did the first upgrade to 1.8.
> >>
> >> Thanks
> >>
> >> Tom
> >>
> >>
> >> -----Original Message-----
> >> From: Lonnie Abelbeck [mailto:[email protected]]
> >> Sent: 22 July 2013 17:23
> >> To: AstLinux Users Mailing List
> >> Cc: Tom Chadwin
> >> Subject: Re: [Astlinux-users] Delay before audio is audible
> >>
> >> Tom,
> >>
> >> Are you setting "directmedia=no" in your sip.conf for the local
extensions
> >> ?
> >> This supersedes the old "canreinvite=no" in Asterisk 1.4 .
> >>
> >> Lonnie
> >>
> >>
> >>
> >> On Jul 22, 2013, at 10:26 AM, Tom Chadwin wrote:
> >>
> >> > Hi Michael
> >> >
> >> > I'm not using the r option in the dial command (we use TtHh). Also,
> >> calls
> >> > via our berofix work correctly - it is only internal SIP calls
(between
> >> > Snom300/320s) which exhibit the behaviour.
> >> >
> >> > We were using Asterisk 1.8 before, whichever version was packaged in
> >> with
> >> > the immediately previous version of Astlinux.
> >> >
> >> > Any other ideas?
> >> >
> >> > Thanks
> >> >
> >> > Tom
> >> >
> >> >
> >> > -----Original Message-----
> >> > From: Michael Keuter [mailto:[email protected]]
> >> > Sent: 22 July 2013 15:57
> >> > To: [email protected]; AstLinux Users Mailing List
> >> > Subject: Re: [Astlinux-users] Delay before audio is audible
> >> >
> >> >
> >> > Am 22.07.2013 um 16:45 schrieb "Tom Chadwin"
> >> > <[email protected]>:
> >> >
> >> > Hello all
> >> >
> >> > We upgraded recently to the most recent 1.8 Astlinux, and we have a
> >> > problem
> >> > we've not encountered before. Intermittently on internal SIP-SIP
calls
> >> > only,
> >> > there is no audio for a varying number of seconds (between 2 and 10)
-
> >> > caller hears no ringing tone, and neither party can hear each other
> >> until
> >> > audio establishes after this delay. The problem has appeared since
the
> >> > latest Astlinux upgrade. I don't see anything untoward on the
console,
> >> but
> >> > I'm no expert. External (DAHDI ISDN) calls are fine.
> >> >
> >> > Does anyone have any ideas, or can point me toward how to debug?
> >> >
> >> > Thanks
> >> >
> >> > Tom
> >> >
> >> >
> >> > I had a similar issue in combination with a Berofix and the dial
command
> >> > using the "r" option.
> >> > The solution was not using the "r" option, and instead activating the
> >> "Early
> >> > Audio" option in the Berofix ("ea=1").
> >> > Check if you are using a Dial with "r" (ringing indication) option
> >> > somewhere.
> >> > Which Asterisk version did you use before the upgrade?
> >> >
> >> > Michael
> >> >
> >> > http://www.mksolutions.info
> >> >
> >> >
> >> >
> >> >
> >> >
> >> >
> >> >
> >>
----------------------------------------------------------------------------
> >> --
> >> > See everything from the browser to the database with AppDynamics
> >> > Get end-to-end visibility with application monitoring from
AppDynamics
> >> > Isolate bottlenecks and diagnose root cause in seconds.
> >> > Start your free trial of AppDynamics Pro today!
> >> >
> >>
> http://pubads.g.doubleclick.net/gampad/clk?id=48808831
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> >> trk
> >> > _______________________________________________
> >> > Astlinux-users mailing list
> >> > [email protected]
> >> > https://lists.sourceforge.net/lists/listinfo/astlinux-users
> >> >
> >> > Donations to support AstLinux are graciously accepted via PayPal to
> >> [email protected].
> >> >
> >> >
> >>
> >>
> >>
> >>
----------------------------------------------------------------------------
--
> >> Get 100% visibility into Java/.NET code with AppDynamics Lite!
> >> It's a free troubleshooting tool designed for production.
> >> Get down to code-level detail for bottlenecks, with <2% overhead.
> >> Download for free and get started troubleshooting in minutes.
> >>
> http://pubads.g.doubleclick.net/gampad/clk?id=48897031
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> trk
> >> _______________________________________________
> >> Astlinux-users mailing list
> >> [email protected]
> >> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> >>
> >> Donations to support AstLinux are graciously accepted via PayPal to
> [email protected].
> >
> >
> >
> >--
> >Kristian Kielhofner
> >
>
>---------------------------------------------------------------------------
---
> >Get 100% visibility into Java/.NET code with AppDynamics Lite!
> >It's a free troubleshooting tool designed for production.
> >Get down to code-level detail for bottlenecks, with <2% overhead.
> >Download for free and get started troubleshooting in minutes.
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> ktrk
> >_______________________________________________
> >Astlinux-users mailing list
> >[email protected]
> >https://lists.sourceforge.net/lists/listinfo/astlinux-users
> >
> >Donations to support AstLinux are graciously accepted via PayPal to
> [email protected].
>
>
>
----------------------------------------------------------------------------
--
> Get 100% visibility into Java/.NET code with AppDynamics Lite!
> It's a free troubleshooting tool designed for production.
> Get down to code-level detail for bottlenecks, with <2% overhead.
> Download for free and get started troubleshooting in minutes.
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<http://pubads.g.doubleclick.net/gampad/clk?id=48897031&iu=/4140/ostg.clk>
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> trk
> _______________________________________________
> Astlinux-users mailing list
> [email protected]
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to
> [email protected].
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