[Serusers] ser: problems with hearing voice

Sat May 27 01:06:35 CEST 2006

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 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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