[OpenSIPS-Users] Problem NAT RTPproxy

[Bogdan-Andrei Iancu]

The relevant one should be INVITE leaving you opensips - to see if 
RTPproxy was inserted in the SDP.

Regards,
Bogdan

On 04/02/2012 11:41 PM, magnusadilsom at gmail.com wrote:
> In ngrep traffic check no active rdp-session-id
>
> but do not know how to solve
>
>
> #
> U +3.135110 IP-ASTERISK:5060 -> IP_OPENSIPS:5060
> INVITE sip:100@ IP_OPENSIPS SIP/2.0
> Via: SIP/2.0/UDP IP-ASTERISK:5060;branch=z9hG4bK3e684698;rport
> Max-Forwards: 70
> From: "3414741468" <sip:TRK00253-001 at IP-ASTERISK>;tag=as33306c2a
> To: <sip:100 at IP_OPENSIPS>
> Contact: <sip:TRK00253-001 at IP-ASTERISK>
> Call-ID: 46ea6e9819e3583c59479d9304cc2b4f at IP-ASTERISK
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 1.6.2.20
> Date: Mon, 26 Mar 2012 16:29:17 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 333
>
> v=0
> o=root 1324806659 1324806659 IN IP4 IP-ASTERISK
> s=Asterisk PBX 1.6.2.20
> c=IN IP4 IP-ASTERISK
> t=0 0
> m=audio 10788 RTP/AVP 0 18 8 3 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
>
>
>
> tanks
>
>
>
>
>
> Bogdan-Andrei Iancu wrote:
>> Well, you know, one is what we want to do , another we actually get.
>>
>> I was rather asking if, making a sip capture (with ngrep) you see in 
>> your call the RTPproxy insertion - check it in traffic, not in script.
>>
>> Regards,
>> Bogdan
>>
>> On 04/02/2012 10:05 PM, magnusadilsom at gmail.com wrote:
>>> hi, yes, rtpproxy is active in invite 200
>>>
>>> onreply_route[3] {
>>>     if ((isflagset(5) || isbflagset(0)) && status =~ 
>>> "(183)|(2[0-9][0-9])" && has_body("application/sdp")) {
>>>         if (rtpproxy_answer()) {
>>>             log("L_INFO: rtpproxy_answer NAT");
>>>         }
>>>     }
>>>     if (!subst_uri('/(sip:.*);nat=yes/\1/')) {
>>>         search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');
>>>     }
>>>     exit;
>>> }
>>>
>>>
>>> But i'm implemented this in invite route
>>>
>>> if (is_method("INVITE") {
>>>      if ($si == "IP ASTERISK" && is_method("INVITE")) {
>>>             fix_nated_contact();
>>>             fix_nated_sdp("1");
>>>             xlog("L_INFO", "NAT detected3 PSTN for SIP");
>>>             setflag(5);
>>>             return;
>>>         }
>>>   }
>>>
>>> and worked, but I think it is not correct
>>>
>>> tansk
>>>
>>>
>>> Bogdan-Andrei Iancu wrote:
>>>> Hi Magnus,
>>>>
>>>> attaching cfg files is useless, as no one will debug the script, 
>>>> but we will help you to debug your script.
>>>>
>>>> So, for the non-working case (PSTN to SIP) does your script force 
>>>> RTPproxy in INVITE and 200 OK ?
>>>>
>>>> Regards,
>>>> Bogdan
>>>>
>>>> On 03/29/2012 01:52 AM, magnusadilsom at gmail.com wrote:
>>>>> I have phones (some behind NAT) connecting to Opensips server an 
>>>>> Asterisk and an rtpproxy as seen below:
>>>>>
>>>>> rtpproxy started with
>>>>> ps -aux | grep rtpproxy
>>>>> root     15666  0.0  0.0  14472   920 ?        Ssl  Mar23   0:05 
>>>>> ./rtpproxy -F -l 189.254.2.19 -s udp:* 7890 -d DBUG LOG_LOCAL3
>>>>>
>>>>>
>>>>>
>>>>> UAC1 username = 
>>>>> 100------------Firewall/router--------------------Opensips 
>>>>> 1.7---------- RTP PROXY------------Asterisk 1.6
>>>>> 192.168.1.10                    192.168.1.1                    
>>>>> 65.254.63.212          189.254.2.19           190.61.201.89
>>>>>                       external ip dinamic 169.254.2.2
>>>>>
>>>>>
>>>>> - Calls between UAC are OK (both SIP and RTP).
>>>>> - Calls UAC for PSTN is OK.
>>>>> - Did numbers is received in Asterisk, and destination for UAC 
>>>>> registered in opensips, but no work audio .
>>>>> (EX User call cellphone for DID 54115368566, call is received in 
>>>>> asterisk, and destination for user 100, registered in opensips)

-- 
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

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