Hi Gabriel,
So you are not using rtpproxy, but rely on the fact that * is all the
time public. In this case, audio from caller to * should work all the
time as the destination is public (of course, if the caller does send
RTP). For the other way around, you can be sure * sends RTP to the
public IP of the NAT (of the client) by doing fix_nated_sdp("1") for the
invite - this will force the COMEDIA support in *.
Anyhow, for RTP nat traversal to work, it is mandatory for the party
behind the nat to start sending RTP (to open the NAT). If the natted
party will send no RTP, there will be no audio at all.
Regard,
Bogdan
Gabriel Bermudez wrote:
> 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:0059389954...@opensips_public_ip SIP/2.0
> Method: INVITE
> [Resent Packet: False]
> Message Header
> To: <sip:0059389954...@opensips_public_ip>
> SIP to address: sip:0059389954...@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:710406...@opensips_public_ip>;tag=41a8c40d
> SIP Display info: "710406702"
> SIP from address: sip:710406...@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-1ee3a...@uac_public_ip
> Max-Forwards: 70
> Contact: <sip:710406...@uac_public_ip:10759>
> Contact Binding: <sip:710406...@uac_public_ip:10759>
> URI: <sip:710406...@uac_public_ip:10759>
> SIP contact address: sip:710406...@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,
>
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>
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