[OpenSIPS-Users] no audio from caller when using nathelper

[Gabriel Bermudez]

Hi everyone,

I'm using the nathelper and dispatcher module to send calls to an 
Asterisk server.  I'm using the Asterisk as a SIP to H.323 converter 
because our PSTN gateway only speaks H.323
For some reason *sometimes* the caller does not send RTP traffic to the 
opensips (one way audio).  The caller's UA is behind a NAT, but it 
doesn't gets detected as a nated UA, so the RTP flow is between the 
client's public IP and the Asterisk public IP (rtpproxy is not used).  
I'm not sure if this problems happens also with UAs that get NAT 
detected (not seen it happen).  I used tshark to capture the invite from 
an undetected NAT UA (changed the UA ip with *uac_public_ip* and 
opensip's ip with *opensips_public_ip*)

Session Initiation Protocol
    Request-Line: INVITE sip:0059389954277 at opensips_public_ip SIP/2.0
        Method: INVITE
        [Resent Packet: False]
    Message Header
        To: <sip:0059389954277 at opensips_public_ip>
            SIP to address: sip:0059389954277 at opensips_public_ip
        Accept: 
application/dtmf-relay,application/sdp,text/plain,message/sipfrag,application/sip
        User-Agent: YV1/1.2.0
        Via: SIP/2.0/UDP uac_public_ip:10759;rport;branch=z9hG4bK474bfa15
            Transport: UDP
            Sent-by Address: uac_public_ip
            Sent-by port: 10759
            RPort: rport
            Branch: z9hG4bK474bfa15
        From: "710406702"<sip:710406702 at opensips_public_ip>;tag=41a8c40d
            SIP Display info: "710406702"
            SIP from address: sip:710406702 at opensips_public_ip
            SIP tag: 41a8c40d
        Allow: UPDATE,INFO,PRACK,REFER,NOTIFY,INVITE,ACK,OPTIONS,BYE,CANCEL
        Allow-Events: refer
        Call-ID: 27f2a0fb-390a1f2a-5e9e57cd-1ee3a7b0 at uac_public_ip
        Max-Forwards: 70
        Contact: <sip:710406702 at uac_public_ip:10759>
            Contact Binding: <sip:710406702 at uac_public_ip:10759>
                URI: <sip:710406702 at uac_public_ip:10759>
                    SIP contact address: sip:710406702 at uac_public_ip:10759
        Session-Expires: 1800
        Content-Length: 313
        Content-Type: application/sdp
        Supported: timer,100rel,join,tdialog,replaces,norefersub,histinfo
        CSeq: 57741 INVITE
            Sequence Number: 57741
            Method: INVITE
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): ipr1B24E8AED4 4453550 4453550 
IN IP4 uac_public_ip
                Owner Username: ipr1B24E8AED4
                Session ID: 4453550
                Session Version: 4453550
                Owner Network Type: IN
                Owner Address Type: IP4
                Owner Address: uac_public_ip
            Session Name (s): -
            Connection Information (c): IN IP4 uac_public_ip
                Connection Network Type: IN
                Connection Address Type: IP4
                Connection Address: uac_public_ip
            Time Description, active time (t): 0 0
                Session Start Time: 0
                Session Stop Time: 0
            Media Description, name and address (m): audio 10760 RTP/AVP 
0 8 4 18 101
                Media Type: audio
                Media Port: 10760
                Media Proto: RTP/AVP
                Media Format: ITU-T G.711 PCMU
                Media Format: ITU-T G.711 PCMA
                Media Format: ITU-T G.723
                Media Format: ITU-T G.729
                Media Format: 101
            Media Attribute (a): rtpmap:0 PCMU/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 0
                MIME Type: PCMU
            Media Attribute (a): rtpmap:8 PCMA/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 8
                MIME Type: PCMA
            Media Attribute (a): rtpmap:4 G723/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 4
                MIME Type: G723
            Media Attribute (a): rtpmap:18 G729/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 18
                MIME Type: G729
            Media Attribute (a): rtpmap:101 telephone-event/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 101
                MIME Type: telephone-event
            Media Attribute (a): ptime:20
                Media Attribute Fieldname: ptime
                Media Attribute Value: 20
            Media Attribute (a): fmtp:101 0-16
                Media Attribute Fieldname: fmtp
                Media Format: 101 [telephone-event]
                Media format specific parameters: 0-16
            Media Attribute (a): fmtp:4 ptime=30;bitrate=6.3
                Media Attribute Fieldname: fmtp
                Media Format: 4 [telephone-event]
                Media format specific parameters: ptime=30
                Media format specific parameters: bitrate=6.3

I really don't find anything wrong with it but I'm no SIP expert.  Can 
some one help me with some pointers.
Thanks for you help.

Regards,