[Serusers] ser: problems with hearing voice

Americania .it americania at hotmail.com
Fri May 26 23:20:04 CEST 2006


I've opened port 8000/8001 on my router ..

I've installed Portrptr to monitor wich ports are used by 3cx Phone (it says 
5062 5063).
I've opened 5060 to 5070 udp) too.
Nothing has changed : no voice.

I've tried XLite : same thing , no voice.

I've tried to make a call from my Lan where Ser is installed to an user 
outside and he can hear me but I can't hear him.

If I enstablish a call from an outside user to another outside user .. no 
voice at all!

I attach X-Lite log for this case:

What can I do now ???



SEND TIME: 3474935
SEND >> 80.105.2.110:5060
INVITE sip:vicky at 80.105.2.110 SIP/2.0
Via: SIP/2.0/UDP 
192.168.0.102:5060;rport;branch=z9hG4bK2C92FDF50A36406697C9FDB48723EC39
From: claudio <sip:claudio at 80.105.2.110>;tag=512156865
To: <sip:vicky at 80.105.2.110>
Contact: <sip:claudio at 192.168.0.102:5060>
Call-ID: 374B53F3-8081-45B1-AD92-4075AF931C00 at 192.168.0.102
CSeq: 21421 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1105x
Content-Length: 308

v=0
o=claudio 3474591 3474934 IN IP4 192.168.0.102
s=X-Lite
c=IN IP4 192.168.0.102
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

RECEIVE TIME: 3475635
RECEIVE << 80.105.2.110:5060
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 
192.168.0.102:5060;rport=39267;branch=z9hG4bK2C92FDF50A36406697C9FDB48723EC39;received=62.101.126.230
From: claudio <sip:claudio at 80.105.2.110>;tag=512156865
To: <sip:vicky at 80.105.2.110>
Call-ID: 374B53F3-8081-45B1-AD92-4075AF931C00 at 192.168.0.102
CSeq: 21421 INVITE
Server: Sip EXpress router (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 10.0.0.133:5060 "Noisy feedback tells:  pid=9102 
req_src_ip=62.101.126.230 req_src_port=39267 in_uri=sip:vicky at 80.105.2.110 
out_uri=sip:vicky at 87.1.193.94:5060 via_cnt==1"


RECEIVE TIME: 3476404
RECEIVE << 80.105.2.110:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 
192.168.0.102:5060;received=62.101.126.230;rport=39267;branch=z9hG4bK2C92FDF50A36406697C9FDB48723EC39
Record-Route: <sip:10.0.0.133;ftag=512156865;lr=on>
From: claudio <sip:claudio at 80.105.2.110>;tag=512156865
To: <sip:vicky at 80.105.2.110>;tag=806313b969ebda11a7f5000d613deeec
Call-ID: 374B53F3-8081-45B1-AD92-4075AF931C00 at 192.168.0.102
CSeq: 21421 INVITE
Contact: <sip:vicky at 87.1.193.94:5060>
Max-Forwards: 16
Server: SIPPER for 3CX Phone
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY
Content-Length: 0


RECEIVE TIME: 3480857
RECEIVE << 80.105.2.110:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
192.168.0.102:5060;received=62.101.126.230;rport=39267;branch=z9hG4bK2C92FDF50A36406697C9FDB48723EC39
Record-Route: <sip:10.0.0.133;ftag=512156865;lr=on>
From: claudio <sip:claudio at 80.105.2.110>;tag=512156865
To: <sip:vicky at 80.105.2.110>;tag=806313b969ebda11a7f5000d613deeec
Call-ID: 374B53F3-8081-45B1-AD92-4075AF931C00 at 192.168.0.102
CSeq: 21421 INVITE
Contact: <sip:vicky at 87.1.193.94:5060>
Max-Forwards: 16
Server: SIPPER for 3CX Phone
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 5424921 5424921 IN IP4 192.168.1.2
s=call
c=IN IP4 10.0.0.133
t=0 0
m=audio 35268 RTP/AVP 0 8 3 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=nortpproxy:yes

SEND TIME: 3480866
SEND >> 10.0.0.133:5060
ACK sip:vicky at 87.1.193.94:5060 SIP/2.0
Via: SIP/2.0/UDP 
192.168.0.102:5060;rport;branch=z9hG4bK28E629C6EC9342FFA3B89AF152782C9D
From: claudio <sip:claudio at 80.105.2.110>;tag=512156865
To: <sip:vicky at 80.105.2.110>;tag=806313b969ebda11a7f5000d613deeec
Contact: <sip:claudio at 192.168.0.102:5060>
Route: <sip:10.0.0.133;ftag=512156865;lr=on>
Call-ID: 374B53F3-8081-45B1-AD92-4075AF931C00 at 192.168.0.102
CSeq: 21421 ACK
Max-Forwards: 70
Content-Length: 0


RECEIVE TIME: 3493387
RECEIVE << 80.105.2.110:5060

SEND TIME: 3500646
SEND >> 80.105.2.110:5060
REGISTER sip:80.105.2.110 SIP/2.0
Via: SIP/2.0/UDP 
192.168.0.102:5060;rport;branch=z9hG4bKC9C44D63BB0344C4B10F04D9C36AD603
From: claudio <sip:claudio at 80.105.2.110>;tag=425214505
To: claudio <sip:claudio at 80.105.2.110>
Contact: "claudio" <sip:claudio at 192.168.0.102:5060>
Call-ID: 63C1074ADB2E448487DECC8994F610F3 at 80.105.2.110
CSeq: 48998 REGISTER
Expires: 160
Max-Forwards: 70
User-Agent: X-Lite release 1105x
Content-Length: 0


SEND TIME: 3502150
SEND >> 80.105.2.110:5060
REGISTER sip:80.105.2.110 SIP/2.0
Via: SIP/2.0/UDP 
192.168.0.102:5060;rport;branch=z9hG4bKC9C44D63BB0344C4B10F04D9C36AD603
From: claudio <sip:claudio at 80.105.2.110>;tag=425214505
To: claudio <sip:claudio at 80.105.2.110>
Contact: "claudio" <sip:claudio at 192.168.0.102:5060>
Call-ID: 63C1074ADB2E448487DECC8994F610F3 at 80.105.2.110
CSeq: 48998 REGISTER
Expires: 160
Max-Forwards: 70
User-Agent: X-Lite release 1105x
Content-Length: 0


RECEIVE TIME: 3502580
RECEIVE << 80.105.2.110:5060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
192.168.0.102:5060;rport=39267;branch=z9hG4bKC9C44D63BB0344C4B10F04D9C36AD603;received=62.101.126.230
From: claudio <sip:claudio at 80.105.2.110>;tag=425214505
To: claudio 
<sip:claudio at 80.105.2.110>;tag=2a08c327b8d597781b526eaf86695180.f298
Call-ID: 63C1074ADB2E448487DECC8994F610F3 at 80.105.2.110
CSeq: 48998 REGISTER
WWW-Authenticate: Digest realm="80.105.2.110", 
nonce="447770198404c4bc9438e9ce6159814a63b777d4"
Server: Sip EXpress router (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 10.0.0.133:5060 "Noisy feedback tells:  pid=9102 
req_src_ip=62.101.126.230 req_src_port=39267 in_uri=sip:80.105.2.110 
out_uri=sip:80.105.2.110 via_cnt==1"


SEND TIME: 3502583
SEND >> 80.105.2.110:5060
REGISTER sip:80.105.2.110 SIP/2.0
Via: SIP/2.0/UDP 
192.168.0.102:5060;rport;branch=z9hG4bKC8C6FD549B75482F9BBE53C7AE958465
From: claudio <sip:claudio at 80.105.2.110>;tag=425214505
To: claudio <sip:claudio at 80.105.2.110>
Contact: "claudio" <sip:claudio at 192.168.0.102:5060>
Call-ID: 63C1074ADB2E448487DECC8994F610F3 at 80.105.2.110
CSeq: 48999 REGISTER
Expires: 160
Authorization: Digest 
username="claudio",realm="80.105.2.110",nonce="447770198404c4bc9438e9ce6159814a63b777d4",response="24188c3172d488f1296a7c2ad2a048d6",uri="sip:80.105.2.110"
Max-Forwards: 70
User-Agent: X-Lite release 1105x
Content-Length: 0


RECEIVE TIME: 3502841
RECEIVE << 80.105.2.110:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
192.168.0.102:5060;rport=39267;branch=z9hG4bKC8C6FD549B75482F9BBE53C7AE958465;received=62.101.126.230
From: claudio <sip:claudio at 80.105.2.110>;tag=425214505
To: claudio 
<sip:claudio at 80.105.2.110>;tag=2a08c327b8d597781b526eaf86695180.24dd
Call-ID: 63C1074ADB2E448487DECC8994F610F3 at 80.105.2.110
CSeq: 48999 REGISTER
Contact: <sip:claudio at 62.101.126.230:39267>;expires=160
Server: Sip EXpress router (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 10.0.0.133:5060 "Noisy feedback tells:  pid=9102 
req_src_ip=62.101.126.230 req_src_port=39267 in_uri=sip:80.105.2.110 
out_uri=sip:80.105.2.110 via_cnt==1"

____________________________________________________________________________
____________________________________________________________________________



>From: "Andrey Kouprianov" <andrey.kouprianov at gmail.com>
>To: serusers at iptel.org
>Subject: Re: [Serusers] ser: problems with hearing voice
>Date: Wed, 24 May 2006 00:24:05 +0700
>
>Yes you do. RTP (for voice and video) protocol normally uses ports
>8000 for media transfer and 8001 for media control. If you are using
>X-lite, then port 8000 is used definitely. A client like eyeBeam, may
>choose from range of ports (i dont really know which exactly, but I
>always see ports in the range of 6000-7000). And Skype, for instance,
>can use ANY port available.
>
>Anyway, you can monitor ports <= 1024 and open the rest. Those are
>well known and they are the target. There are some ports > 1024 that
>Windows uses for it's services and you might want to find our what
>those (because, i dont really know :) are and monitor them as well.
>
>Good luck.
>
>On 5/23/06, Americania .it <americania at hotmail.com> wrote:
>>Hi,
>>I can' hear any voice  when I call from a pc outside the Lan where ser is
>>installed (I've  got a router-firewall): I've port 5060 UDP/TCP forwarded 
>>to
>>my ser server .
>>Do I have to open other ports ?
>>
>>Thanks
>>
>>
>>_______________________________________________
>>Serusers mailing list
>>Serusers at lists.iptel.org
>>http://lists.iptel.org/mailman/listinfo/serusers
>>
>_______________________________________________
>Serusers mailing list
>Serusers at lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers





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