Re: [OpenSIPS-Users] One way audio with AudioCodes Mediant 2000 and NAT

2009-02-16 Thread Bogdan-Andrei Iancu
Hi Julian, You can still handle the NAT wih COMEDIA even for T.38, but you have to handle the re-INVITE also . In your scenario, who is generating the re-INVITE? Regards, Bogdan Julian Yap wrote: The full story is that I was looking to get T.38 working behind NAT. Unfortunately, no matter

Re: [OpenSIPS-Users] One way audio with AudioCodes Mediant 2000 and NAT

2009-02-16 Thread Julian Yap
In an example scenario, the re-INVITE is handled by the end device. So: PSTN Fax -- GW -- OpenSIPS -- UA (ATA attached to Fax machine) UA answers the call and then sends the re-INVITE which is correct as that is the terminating side. I read this RFC

Re: [OpenSIPS-Users] One way audio with AudioCodes Mediant 2000 and NAT

2009-02-15 Thread Bogdan-Andrei Iancu
Hi Julian, That is cool - in this way you save a lot of bandwidth and processing power with media relaying. Regards, Bogdan Julian Yap wrote: Hi all, I eventually played around with the Audiocodes box and enabled some settings so it worked with Comedia support. Thanks, Julian On

Re: [OpenSIPS-Users] One way audio with AudioCodes Mediant 2000 and NAT

2009-02-15 Thread Julian Yap
The full story is that I was looking to get T.38 working behind NAT. Unfortunately, no matter what I tried, it wouldn't work behind NAT. I had the initial INVITE (G.711) working fine but when there was the T.38 re-INVITE, the RTP media would connect up fine but just wouldn't negotiate properly

Re: [OpenSIPS-Users] One way audio with AudioCodes Mediant 2000 and NAT

2009-02-10 Thread Iñaki Baz Castillo
2009/2/10 Johansson Olle E o...@edvina.net: If both devices are on private IP's, there's going to be two RTP proxys involved if they're on different SIP networks. Each SIP service needs an RTP proxy for supporting their local users. Hi, I don't fully agree on it: alice --- (NAT A) ---

Re: [OpenSIPS-Users] One way audio with AudioCodes Mediant 2000 and NAT

2009-02-10 Thread Johansson Olle E
10 feb 2009 kl. 13.10 skrev Iñaki Baz Castillo: 2009/2/10 Johansson Olle E o...@edvina.net: If both devices are on private IP's, there's going to be two RTP proxys involved if they're on different SIP networks. Each SIP service needs an RTP proxy for supporting their local users. Hi, I

Re: [OpenSIPS-Users] One way audio with AudioCodes Mediant 2000 and NAT

2009-02-10 Thread Iñaki Baz Castillo
2009/2/10 Johansson Olle E o...@edvina.net: alice --- (NAT A) --- ProxyA RtpProxyA --- ProxyB RtpProxyB --- (NAT B) --- bob In this case, when alice calls bob, ProxyA will apply RtpProxyA so the SDP will contain a public IP. Since ProxyB knows that bob is registered behind NAT, it will

Re: [OpenSIPS-Users] One way audio with AudioCodes Mediant 2000 and NAT

2009-02-10 Thread Johansson Olle E
10 feb 2009 kl. 13.44 skrev Iñaki Baz Castillo: 2009/2/10 Johansson Olle E o...@edvina.net: alice --- (NAT A) --- ProxyA RtpProxyA --- ProxyB RtpProxyB --- (NAT B) --- bob In this case, when alice calls bob, ProxyA will apply RtpProxyA so the SDP will contain a public IP. Since ProxyB

Re: [OpenSIPS-Users] One way audio with AudioCodes Mediant 2000 and NAT

2009-02-10 Thread Iñaki Baz Castillo
2009/2/10 Johansson Olle E o...@edvina.net: No, that's not an excuse. RtpProxy must be applied in both the request and response. If a PSTN gateway calls a SIP user behind NAT, you must apply RtpProxy even if the incoming SDP has public address. Not in the INVITE sdp - the device behind the

Re: [OpenSIPS-Users] One way audio with AudioCodes Mediant 2000 and NAT

2009-02-10 Thread Bogdan-Andrei Iancu
Hi Olle, Johansson Olle E wrote: 10 feb 2009 kl. 12.25 skrev Iñaki Baz Castillo: 2009/2/10 julianok...@gmail.com: You don't know if RtpProxy should be running, does it mean you are trying to use it or not? I don't want to spend time inspecting what you want to do by reading your config,

Re: [OpenSIPS-Users] One way audio with AudioCodes Mediant 2000 and NAT

2009-02-10 Thread Julian Yap
Thanks all. I'll check to see if the AudioCodes gateway does have comedia support. That clarifies some half baked NAT/RTP knowledge in my head. - Julian On 2/10/09, Bogdan-Andrei Iancu bog...@voice-system.ro wrote: Hi Olle, Johansson Olle E wrote: 10 feb 2009 kl. 12.25 skrev Iñaki Baz