This looks normal to me. A trace from the AstLinux side showing both legs
(and in pcap format - not just text) would really help.
On Mon, Aug 19, 2013 at 12:49 PM, Tom Chadwin <
[email protected]> wrote:
> 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&iu=/4140/ostg.clk
> > >> 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&iu=/4140/ostg.clk
> > 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.
> > >http://pubads.g.doubleclick.net/gampad/clk?id=48897031&iu=/4140/ostg.cl
> > 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.
> > http://pubads.g.doubleclick.net/gampad/clk?id=48897031&iu=/4140/ostg.clk
> > 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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> _______________________________________________
> 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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> analysis and resources for efficient Application Performance Management.
> Visit us today!
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> [email protected]
> https://lists.sourceforge.net/lists/listinfo/astlinux-users
>
> Donations to support AstLinux are graciously accepted via PayPal to
> [email protected].
>
--
Kristian Kielhofner
------------------------------------------------------------------------------
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