[Freeswitch-users] FS 1.0.6 on EC2 - no audio on outgoing calls
paul gore
paul.gore.j at gmail.com
Thu Jul 8 16:31:22 PDT 2010
I got ngrep trace for port 5060 while making a call to a US number via
siptraffic.com from X-Lite phone. I canceled the call after ~ 20 sec., I
heard no audio not even ringing.
Is there anything in this trace which can help identify the problem?
10.194.206.102:5060 - is my local EC2 IP
184.72.206.204:5060 - is my public EC2 IP
77.72.169.128:5060 - siptraffic.com proxy IP
Thanks!
67.33.160.119:18294 -> 10.194.206.102:5060
INVITE sip:45517705678570 at myserver.com
<sip%3A45517705678570 at myserver.com>SIP/2.0..Via: SIP/2.0/UDP
192.168.0.8:29486
;branch=z9hG4bK-d87543-f
524431af92cef56-1--d87543-;rport..Max-Forwards: 70..Contact: <
sip:4000002 at 67.33.160.119:18027>..To: "45517
709248570"<sip:45517705678570 at myserver.com<sip%3A45517705678570 at myserver.com>>..From:
"4000002"<sip:4000002 at myserver.com <sip%3A4000002 at myserver.com>
>;tag=5f1ec15f..Call-
ID: ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 INVITE..Allow:
INVITE, ACK, CANCEL, OPTIONS, BYE
, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type:
application/sdp..Proxy-Authorization: Digest user
name="4000002",realm="myserver.com
",nonce="cf9019cc-f44a-4568-97d1-e9883fb1821f",uri="sip:45517705678570 at v
ersafon.com
",response="57da72527524e0e065c9a3221bfadd38",cnonce="140f655ff3427f6ba3767ab7040231f3",nc=0000
0001,qop=auth,algorithm=MD5..User-Agent: X-Lite release 1011s stamp
41150..Content-Length: 417....v=0..o=-
8 2 IN IP4 192.168.0.8..s=CounterPath X-Lite 3.0..c=IN IP4
192.168.0.8..t=0 0..m=audio 46298 RTP/AVP 107
119 100 106 0 105 98 8 101..a=alt:1 1 : tomYv1/D Yont/s+3 192.168.0.8
46298..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..
#
U 10.194.206.102:5060 -> 67.33.160.119:18294
SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.0.8:29486
;branch=z9hG4bK-d87543-f524431af92cef56-1--d87543-;r
port=18294;received=67.33.160.119..From: "4000002" <
sip:4000002 at myserver.com <sip%3A4000002 at myserver.com>>;tag=5f1ec15f..To:
"455177092
48570" <sip:45517705678570 at myserver.com<sip%3A45517705678570 at myserver.com>>..Call-ID:
ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 I
NVITE..User-Agent: myserver..Content-Length: 0....
#
U 10.194.206.102:5080 -> 77.72.169.128:5060
INVITE sip:0017705678570 at sip.siptraffic.com<sip%3A0017705678570 at sip.siptraffic.com>SIP/2.0..Via:
SIP/2.0/UDP 184.72.206.204:5080
;rport;branch=z9h
G4bKBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" <
sip:0014444295793 at 184.72.206.204 <sip%3A0014444295793 at 184.72.206.204>
>;tag=18853e82KDe7j.
.To: <sip:0017705678570 at sip.siptraffic.com<sip%3A0017705678570 at sip.siptraffic.com>>..Call-ID:
37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647
2 INVITE..Contact: <sip:gw+voicetrading.com at 184.72.206.204:5080
;transport=udp;gw=voicetrading.com>..User-A
gent: myserver..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE,
INFO, REGISTER, REFER, NOTIFY..
Supported: timer, precondition, path, replaces..Allow-Events: talk,
refer..Content-Type: application/sdp..
Content-Disposition: session..Content-Length: 295..X-FS-Support:
update_display..Remote-Party-ID: "4000002
" <sip:0014444295793 at 184.72.206.204
<sip%3A0014444295793 at 184.72.206.204>>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH
1278518039
1278518040 IN IP4 184.72.206.204..s=FreeSWITCH..c=IN IP4
184.72.206.204..t=0 0..m=audio 31564 RTP/AVP 0 8
3 101 13..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:3
GSM/8000..a=rtpmap:101 telephone-event/80
00..a=fmtp:101 0-16..a=rtpmap:13 CN/8000..a=ptime:20..
#
U 77.72.169.128:5060 -> 10.194.206.102:5080
SIP/2.0 183 Session progress..Via: SIP/2.0/UDP 184.72.206.204:5080
;rport;branch=z9hG4bKBU626KBp16t5Q..From
: "4000002" <sip:0014444295793 at 184.72.206.204<sip%3A0014444295793 at 184.72.206.204>>;tag=18853e82KDe7j..To:
<sip:0017705678570 at sip.siptraffic.co <sip%3A0017705678570 at sip.siptraffic.co>
m>;tag=20113ac4c230cd6412168..Contact:
sip:0017705678570 at 77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c
381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip
Registrar/Proxy Server)..Allow: ACK,BYE,C
ANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE..Content-Type:
application/sdp..Content-Length: 198....v=0..o=C
ARRIER 1278549617 1278549617 IN IP4 77.72.168.40..s=SIP Call..c=IN IP4
77.72.168.40..t=0 0..m=audio 57672
RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101
telephone-event/8000..a=ptime:20..
#
U 10.194.206.102:5060 -> 67.33.160.119:18294
SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP 192.168.0.8:29486
;branch=z9hG4bK-d87543-f524431af92cef56-1-
-d87543-;rport=18294;received=67.33.160.119..From: "4000002" <
sip:4000002 at myserver.com <sip%3A4000002 at myserver.com>>;tag=5f1ec15f..To:
"45517705678570"
<sip:45517705678570 at myserver.com<sip%3A45517705678570 at myserver.com>>;tag=BXgB1FZBUZ3Da..Call-ID:
ZDAzODE0Y2JkZjYzODE5NmVmNjk
zMjg5YzkwMTdkYmY...CSeq: 2 INVITE..Contact:
<sip:45517705678570 at 184.72.206.204:5060;transport=udp>..User-A
gent: myserver..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL,
OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer,
precondition, path, replaces..Allow-Events:
talk, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-
summary, refer..Content-Type: application/sdp..Content-Disposition:
session..Content-Length: 251..Remote-P
arty-ID: "45517705678570"
<sip:45517705678570 at 10.194.206.102<sip%3A45517705678570 at 10.194.206.102>
>;party=calling;privacy=off;screen=no....v=0..
o=FreeSWITCH 1278530815 1278530816 IN IP4
184.72.206.204..s=FreeSWITCH..c=IN IP4 184.72.206.204..t=0 0..m=
audio 18788 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101
telephone-event/8000..a=fmtp:101 0-16..a=sil
enceSupp:off - - - -..a=ptime:20..
#
U 77.72.169.128:5060 -> 10.194.206.102:5080
SIP/2.0 183 Session progress..Via: SIP/2.0/UDP 184.72.206.204:5080
;rport;branch=z9hG4bKBU626KBp16t5Q..From
: "4000002" <sip:0014444295793 at 184.72.206.204<sip%3A0014444295793 at 184.72.206.204>>;tag=18853e82KDe7j..To:
<sip:0017705678570 at sip.siptraffic.co <sip%3A0017705678570 at sip.siptraffic.co>
m>;tag=20113ac4c230cd6412168..Contact:
sip:0017705678570 at 77.72.169.128:5060..Call-ID: 37f59333-04cc-122e-c
381-12313b06cd32..CSeq: 133156472 INVITE..Server: (Very nice Sip
Registrar/Proxy Server)..Allow: ACK,BYE,C
ANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE..Content-Type:
application/sdp..Content-Length: 204....v=0..o=C
ARRIER 1278549619 1278549619 IN IP4 208.167.230.118..s=SIP Call..c=IN IP4
208.167.230.118..t=0 0..m=audio
57786 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101
telephone-event/8000..a=ptime:20..
#
U 67.33.160.119:18294 -> 10.194.206.102:5060
....
#
U 67.33.160.119:18294 -> 10.194.206.102:5060
CANCEL sip:45517705678570 at myserver.com
<sip%3A45517705678570 at myserver.com>SIP/2.0..Via: SIP/2.0/UDP
192.168.0.8:29486
;branch=z9hG4bK-d87543-f
524431af92cef56-1--d87543-;rport..To: "45517705678570"<
sip:45517705678570 at myserver.com <sip%3A45517705678570 at myserver.com>>..From:
"4000002"<s
ip:4000002 at myserver.com <ip%3A4000002 at myserver.com>>;tag=5f1ec15f..Call-ID:
ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYmY...CSeq: 2 CANC
EL..Proxy-Authorization: Digest username="4000002",realm="myserver.com
",nonce="cf9019cc-f44a-4568-97d1-e98
83fb1821f",uri="sip:45517705678570 at myserver.com<sip%3A45517705678570 at myserver.com>
",response="46c7e289f7490c807565c561699b03d6",cnonce="a226c
55446c605ee229f045602b29135",nc=00000002,qop=auth,algorithm=MD5..User-Agent:
X-Lite release 1011s stamp 41
150..Content-Length: 0....
#
U 10.194.206.102:5060 -> 67.33.160.119:18294
SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.0.8:29486
;branch=z9hG4bK-d87543-f524431af92cef56-1--d87543-;rport
=18294;received=67.33.160.119..From: "4000002"
<sip:4000002 at myserver.com<sip%3A4000002 at myserver.com>>;tag=5f1ec15f..To:
"4551770567857
0" <sip:45517705678570 at myserver.com
<sip%3A45517705678570 at myserver.com>>;tag=BXgB1FZBUZ3Da..Call-ID:
ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkY
mY...CSeq: 2 CANCEL..Content-Length: 0....
#
U 10.194.206.102:5060 -> 67.33.160.119:18294
SIP/2.0 487 Request Terminated..Via: SIP/2.0/UDP 192.168.0.8:29486
;branch=z9hG4bK-d87543-f524431af92cef56-
1--d87543-;rport=18294;received=67.33.160.119..From: "4000002" <
sip:4000002 at myserver.com <sip%3A4000002 at myserver.com>>;tag=5f1ec15f..To
: "45517705678570"
<sip:45517705678570 at myserver.com<sip%3A45517705678570 at myserver.com>>;tag=BXgB1FZBUZ3Da..Call-ID:
ZDAzODE0Y2JkZjYzODE5NmVmN
jkzMjg5YzkwMTdkYmY...CSeq: 2 INVITE..User-Agent: myserver..Allow: INVITE,
ACK, BYE, CANCEL, OPTIONS, MESSA
GE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported:
timer, precondition, path, repla
ces..Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
include-session-description, presen
ce.winfo, message-summary, refer..Content-Length: 0....
#
U 10.194.206.102:5080 -> 77.72.169.128:5060
CANCEL sip:0017705678570 at sip.siptraffic.com<sip%3A0017705678570 at sip.siptraffic.com>SIP/2.0..Via:
SIP/2.0/UDP 184.72.206.204:5080
;rport;branch=z9h
G4bKBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" <
sip:0014444295793 at 184.72.206.204 <sip%3A0014444295793 at 184.72.206.204>
>;tag=18853e82KDe7j.
.To: <sip:0017705678570 at sip.siptraffic.com<sip%3A0017705678570 at sip.siptraffic.com>>..Call-ID:
37f59333-04cc-122e-c381-12313b06cd32..CSeq: 13315647
2 CANCEL..Reason: Q.850;cause=16;text="NORMAL_CLEARING"..Content-Length:
0....
#
U 67.33.160.119:18294 -> 10.194.206.102:5060
ACK sip:45517705678570 at myserver.com
<sip%3A45517705678570 at myserver.com>SIP/2.0..Via: SIP/2.0/UDP
192.168.0.8:29486
;branch=z9hG4bK-d87543-f524
431af92cef56-1--d87543-;rport..To: "45517705678570" <
sip:45517705678570 at myserver.com <sip%3A45517705678570 at myserver.com>
>;tag=BXgB1FZBUZ3Da..F
rom: "4000002"<sip:4000002 at myserver.com
<sip%3A4000002 at myserver.com>>;tag=5f1ec15f..Call-ID:
ZDAzODE0Y2JkZjYzODE5NmVmNjkzMjg5YzkwMTdkYm
Y...CSeq: 2 ACK..Content-Length: 0....
#
U 77.72.169.128:5060 -> 10.194.206.102:5080
SIP/2.0 200 Ok..Via: SIP/2.0/UDP
184.72.206.204:5080;rport;branch=z9hG4bKBU626KBp16t5Q..From:
"4000002" <s
ip:0014444295793 at 184.72.206.204
<ip%3A0014444295793 at 184.72.206.204>>;tag=18853e82KDe7j..To:
<sip:0017705678570 at sip.siptraffic.com<sip%3A0017705678570 at sip.siptraffic.com>>..Contact:
s
ip:0017705678570 at 77.72.169.128:5060..Call-ID:
37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 CANCEL
..Server: (Very nice Sip Registrar/Proxy Server)..Allow:
ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSA
GE..Content-Length: 0....
#
U 77.72.169.128:5060 -> 10.194.206.102:5080
SIP/2.0 487 Request terminated..Via: SIP/2.0/UDP 184.72.206.204:5080
;rport;branch=z9hG4bKBU626KBp16t5Q..Fr
om: "4000002"
<sip:0014444295793 at 184.72.206.204<sip%3A0014444295793 at 184.72.206.204>>;tag=18853e82KDe7j..To:
<sip:0017705678570 at sip.siptraffic.
com>..Contact: sip:0017705678570 at 77.72.169.128:5060..Call-ID:
37f59333-04cc-122e-c381-12313b06cd32..CSeq:
133156472 INVITE..Server: (Very nice Sip Registrar/Proxy Server)..Allow:
ACK,BYE,CANCEL,INVITE,REGISTER,OP
TIONS,INFO,MESSAGE..Content-Length: 0....
#
U 10.194.206.102:5080 -> 77.72.169.128:5060
ACK sip:0017705678570 at sip.siptraffic.com<sip%3A0017705678570 at sip.siptraffic.com>SIP/2.0..Via:
SIP/2.0/UDP 184.72.206.204:5080
;rport;branch=z9hG4b
KBU626KBp16t5Q..Max-Forwards: 69..From: "4000002" <
sip:0014444295793 at 184.72.206.204 <sip%3A0014444295793 at 184.72.206.204>
>;tag=18853e82KDe7j..To
: <sip:0017705678570 at sip.siptraffic.com<sip%3A0017705678570 at sip.siptraffic.com>>..Call-ID:
37f59333-04cc-122e-c381-12313b06cd32..CSeq: 133156472 A
CK..Content-Length: 0....
#
On Wed, Jul 7, 2010 at 5:50 PM, paul gore <paul.gore.j at gmail.com> wrote:
> Seems like siptraffic uses 6 ip addresses for media, can that be the
> problem? Is there any setting in a gateway config xml which helps with
> that?
> I will do ngrep thing and update.
>
> On 7/7/10, paul gore <paul.gore.j at gmail.com> wrote:
> > This provider does work on another box which is not natted as ec2.
> > Most puzzling here though is why call originaion via api even not
> > going via siptraffic still gets no audio.
> >
> > On 7/7/10, Tony Graziano <tgraziano at myitdepartment.net> wrote:
> >> You should try from a standalone or local installation to ensure it
> works
> >> with this provider and your account before you attempt to run it on ec2
> >> (imo).
> >>
> >> On Wed, Jul 7, 2010 at 4:16 PM, Sergey Okhapkin
> >> <sos at sokhapkin.dyndns.org>wrote:
> >>
> >>> What "doesn't work" means? It could be (and most likely is not)
> >>> FS-related
> >>> problem
> >>>
> >>> On Wednesday 07 July 2010, Madovsky wrote:
> >>> > I had same problem from this provider without to explain why.
> >>> > One day it works, another day it doesn't, their support is crap...
> >>> >
> >>> > ----- Original Message -----
> >>> > From: Anthony Minessale
> >>> > To: freeswitch-users at lists.freeswitch.org
> >>> > Sent: Wednesday, July 07, 2010 2:37 PM
> >>> > Subject: Re: [Freeswitch-users] FS 1.0.6 on EC2 - no audio on
> >>> > outgoing
> >>> > calls
> >>> >
> >>> >
> >>> > not really, not with so little information.
> >>> >
> >>> >
> >>> >
> >>> > On Wed, Jul 7, 2010 at 1:30 PM, paul gore <paul.gore.j at gmail.com>
> >>> wrote:
> >>> >
> >>> > Firewall is configured according to the wiki, I also tried to
> open
> >>> all
> >>> > udp ports, issue persists.
> >>> > Actually the problem became more complex - outgoing calls don't
> >>> > work
> >>> > with one particular termination provider, siptraffic.com , any
> >>> > ideas
> >>> > why?
> >>> > Outgoing calls also don't work when originating a call via js
> >>> > script
> >>> > or via FS api. Any clues on that one?
> >>> >
> >>> > On 7/6/10, paul gore <paul.gore.j at gmail.com> wrote:
> >>> > > Hi there,
> >>> > > I am experimenting with FS on EC2, I like results, but stuck on
> >>> weird
> >>> > > audio issue - I followed FreeSwitch EC2 wiki article and
> >>> > modified
> >>> > > internal profile
> >>> > > and vars.xml accordingly, but unfortunately still cannot get it
> >>> > > working. Incoming and outgoing calls made using a SIP phone to
> >>> > FS
> >>> > > extensions work just fine. As well as calls to FS from PSTN.
> But
> >>> > > calls to PSTN via gateways result in no audio at all, no ring,
> >>> > > nothing, SIP signaling goes through OK. Sofia status profile
> >>> > shows
> >>> > > correct values for Ext-RTP-IP for both profiles -
> >>> > > my static public IP, RTP-IP shows local IP.
> >>> > > Any thoughts on that? Anybody can share working profile
> >>> configuration
> >>> > > may be?
> >>> > > Please help, I really need to get this going.
> >>> > >
> >>> > > Thanks.
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> > UNSUBSCRIBE:
> >>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>> >
> >>> >
> >>> >
> >>> > FreeSWITCH http://www.freeswitch.org/
> >>> > ClueCon http://www.cluecon.com/
> >>> > Twitter: http://twitter.com/FreeSWITCH_wire
> >>> >
> >>> > AIM: anthm
> >>> >
> >>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> <MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> >>> >
> >>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> <PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> >>> > IRC: irc.freenode.net #freeswitch
> >>> >
> >>> > FreeSWITCH Developer Conference
> >>> >
> >>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> <sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> >>> >
> >>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> <googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> >>> > pstn:+19193869900
> >>> >
> >>> >
> >>> >
> >>> >
> >>>
> ---------------------------------------------------------------------------
> >>> > ---
> >>> >
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> > UNSUBSCRIBE:
> >>> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >>
> >> --
> >> ======================
> >> Tony Graziano, Manager
> >> Telephone: 434.984.8430
> >> sip: tgraziano at voice.myitdepartment.net
> >> Fax: 434.984.8431
> >>
> >> Email: tgraziano at myitdepartment.net
> >>
> >> LAN/Telephony/Security and Control Systems Helpdesk:
> >> Telephone: 434.984.8426
> >> sip: helpdesk at voice.myitdepartment.net
> >> Fax: 434.984.8427
> >>
> >> Helpdesk Contract Customers:
> >> http://www.myitdepartment.net/gethelp/
> >>
> >> Why do mathematicians always confuse Halloween and Christmas?
> >> Because 31 Oct = 25 Dec.
> >>
> >
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100708/2727d960/attachment-0001.html
More information about the FreeSWITCH-users
mailing list