23 May
2006
23 May
'06
1:24 p.m.
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(a)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(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
26 May
26 May
5:20 p.m.
New subject: [Serusers] ser: problems with hearing voice
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@80.105.2.110 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.102:5060;rport;branch=z9hG4bK2C92FDF50A36406697C9FDB48723EC39
From: claudio <sip:claudio@80.105.2.110>;tag=512156865
To: <sip:vicky@80.105.2.110>
Contact: <sip:claudio@192.168.0.102:5060>
Call-ID: 374B53F3-8081-45B1-AD92-4075AF931C00(a)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@80.105.2.110>;tag=512156865
To: <sip:vicky@80.105.2.110>
Call-ID: 374B53F3-8081-45B1-AD92-4075AF931C00(a)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@80.105.2.110
out_uri=sip:vicky@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@80.105.2.110>;tag=512156865
To: <sip:vicky@80.105.2.110>;tag=806313b969ebda11a7f5000d613deeec
Call-ID: 374B53F3-8081-45B1-AD92-4075AF931C00(a)192.168.0.102
CSeq: 21421 INVITE
Contact: <sip:vicky@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@80.105.2.110>;tag=512156865
To: <sip:vicky@80.105.2.110>;tag=806313b969ebda11a7f5000d613deeec
Call-ID: 374B53F3-8081-45B1-AD92-4075AF931C00(a)192.168.0.102
CSeq: 21421 INVITE
Contact: <sip:vicky@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@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@80.105.2.110>;tag=512156865
To: <sip:vicky@80.105.2.110>;tag=806313b969ebda11a7f5000d613deeec
Contact: <sip:claudio@192.168.0.102:5060>
Route: <sip:10.0.0.133;ftag=512156865;lr=on>
Call-ID: 374B53F3-8081-45B1-AD92-4075AF931C00(a)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@80.105.2.110>;tag=425214505
To: claudio <sip:claudio@80.105.2.110>
Contact: "claudio" <sip:claudio@192.168.0.102:5060>
Call-ID: 63C1074ADB2E448487DECC8994F610F3(a)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@80.105.2.110>;tag=425214505
To: claudio <sip:claudio@80.105.2.110>
Contact: "claudio" <sip:claudio@192.168.0.102:5060>
Call-ID: 63C1074ADB2E448487DECC8994F610F3(a)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@80.105.2.110>;tag=425214505
To: claudio
<sip:claudio@80.105.2.110>;tag=2a08c327b8d597781b526eaf86695180.f298
Call-ID: 63C1074ADB2E448487DECC8994F610F3(a)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@80.105.2.110>;tag=425214505
To: claudio <sip:claudio@80.105.2.110>
Contact: "claudio" <sip:claudio@192.168.0.102:5060>
Call-ID: 63C1074ADB2E448487DECC8994F610F3(a)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@80.105.2.110>;tag=425214505
To: claudio
<sip:claudio@80.105.2.110>;tag=2a08c327b8d597781b526eaf86695180.24dd
Call-ID: 63C1074ADB2E448487DECC8994F610F3(a)80.105.2.110
CSeq: 48999 REGISTER
Contact: <sip:claudio@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(a)gmail.com>
To: serusers(a)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(a)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(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
_______________________________________________
Serusers mailing list
Serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers 
7:06 p.m.
New subject: [Serusers] ser: problems with hearing voice
Hi,
can you draw your network scenario (just where is SER, SIP phones and
NAT devices in the network) ? It'll be easier to see the problem.
Since you run SER behind NAT (am I right?) nathelper checks for private
addresses {RFC1918) and so the logic of ser.cfg is not then appropriate.
Can you send your ser.cfg? Have you read at onsip.org the getting
started document?
Next time catch your SIP messages on the server by:
ngrep -d any -W byline -O /tmp/trace.log port 5060
or
tcpdump -p -s 0 -i any -e port 5060 -w /tmp/trace.log
Ladislav
Americania .it wrote:
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@80.105.2.110 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.102:5060;rport;branch=z9hG4bK2C92FDF50A36406697C9FDB48723EC39
From: claudio <sip:claudio@80.105.2.110>;tag=512156865
To: <sip:vicky@80.105.2.110>
Contact: <sip:claudio@192.168.0.102:5060>
Call-ID: 374B53F3-8081-45B1-AD92-4075AF931C00(a)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@80.105.2.110>;tag=512156865
To: <sip:vicky@80.105.2.110>
Call-ID: 374B53F3-8081-45B1-AD92-4075AF931C00(a)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@80.105.2.110 out_uri=sip:vicky@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@80.105.2.110>;tag=512156865
To: <sip:vicky@80.105.2.110>;tag=806313b969ebda11a7f5000d613deeec
Call-ID: 374B53F3-8081-45B1-AD92-4075AF931C00(a)192.168.0.102
CSeq: 21421 INVITE
Contact: <sip:vicky@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@80.105.2.110>;tag=512156865
To: <sip:vicky@80.105.2.110>;tag=806313b969ebda11a7f5000d613deeec
Call-ID: 374B53F3-8081-45B1-AD92-4075AF931C00(a)192.168.0.102
CSeq: 21421 INVITE
Contact: <sip:vicky@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@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@80.105.2.110>;tag=512156865
To: <sip:vicky@80.105.2.110>;tag=806313b969ebda11a7f5000d613deeec
Contact: <sip:claudio@192.168.0.102:5060>
Route: <sip:10.0.0.133;ftag=512156865;lr=on>
Call-ID: 374B53F3-8081-45B1-AD92-4075AF931C00(a)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@80.105.2.110>;tag=425214505
To: claudio <sip:claudio@80.105.2.110>
Contact: "claudio" <sip:claudio@192.168.0.102:5060>
Call-ID: 63C1074ADB2E448487DECC8994F610F3(a)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@80.105.2.110>;tag=425214505
To: claudio <sip:claudio@80.105.2.110>
Contact: "claudio" <sip:claudio@192.168.0.102:5060>
Call-ID: 63C1074ADB2E448487DECC8994F610F3(a)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@80.105.2.110>;tag=425214505
To: claudio
<sip:claudio@80.105.2.110>;tag=2a08c327b8d597781b526eaf86695180.f298
Call-ID: 63C1074ADB2E448487DECC8994F610F3(a)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@80.105.2.110>;tag=425214505
To: claudio <sip:claudio@80.105.2.110>
Contact: "claudio" <sip:claudio@192.168.0.102:5060>
Call-ID: 63C1074ADB2E448487DECC8994F610F3(a)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@80.105.2.110>;tag=425214505
To: claudio
<sip:claudio@80.105.2.110>;tag=2a08c327b8d597781b526eaf86695180.24dd
Call-ID: 63C1074ADB2E448487DECC8994F610F3(a)80.105.2.110
CSeq: 48999 REGISTER
Contact: <sip:claudio@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(a)gmail.com>
To: serusers(a)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(a)hotmail.com> wrote:
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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
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28 May
28 May
6:24 a.m.
New subject: [Serusers] ser: problems with hearing voice
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 ???
First, you should use module nathelper in your ser. It will help you to fix
intranet IP address problem. Second, check if you can send and recieve RTP
data in range 2000 - 6000, because in you scenario 200 OK send audio from
port 35268 (of course it's a random)
m=audio 35268 RTP/AVP 0 8 3 97 101
^^^^^
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3 comments
4 participants
participants (4)
-
Alexandr Dubovikov -
Americania .it -
Andrey Kouprianov -
Ladislav Andel