Whether it is IAX, SIP, H323 or ?????
These are authentication handshakes to establish an rtp stream. SIP = user name and password in a standardized IP packet IAX = same H.323 = same Is also has to do with what codec are supported as well. As far as NAT is concerned! Yep, tell your ISP to forward the authentication port or just junk their gear and get something like a low end Cisco. Or Get IP Phones with STUN (a little pricey) Or Trick!!!! Use some type of tunneling gear to an outside IP (outside your NAT) and then bounce your authentication from this new gateway!!! i.e. establish a VPN connection to an outside router from an internal router and drive the call through there. Brad _____ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of A. Levy Sent: Tuesday, March 27, 2007 6:54 AM To: [email protected] Subject: [asterisk-users] Re: Question about DSP in Digium card well, ...,we did not choose SIP because our customers are located behind NAT router (using private IP's) and those routers are not managed by them but by the ISP so it is very difficult to establish full duplex phone calls because you can not initiate voice over ip session from the internet (outside) to LAN side (inside) with private IP's. We could not establish 2-way phone calls, I mean, the conversation is listened in 1-way only. As I mentioned before, we can not configure PAT into the NAT router neither because is handled by the ISP and the passwords are unknown .... That's why we decided to use IAX instead of SIP, I mean, IAX is more robust than SIP when the NAT router is 3th-party managed and the PAT feature is not enable. On the other and we tested IAX over dialup links and it worked fine.... Those are the reasons we choose IAX as "acess protocol" to our SIP/H323 Network. You know, the access networks of the customers are different completely: Private IP Address over DSL lines (NAT Router), Public IP Address over DSL lines, Corporate Networks over dedicated Links (Public and IP Addresses), Dialup links, ...... Any comment would be welcomed, thanks a lot Levy.- 2007/3/24, A. Levy <[EMAIL PROTECTED]>: Hello. I have a TE405P Digium Card (4 E1's) with ISDN protocol and I need to find out if there is any limitation about DSP capabilities, I mean, I am not sure how many phone calls my Digium card supports, simultaneously. The calling flow goes from IAX <-> ISDN. I am running this card into CPU like this: - Micro PIV 3.0 - 1Gbyte Memory Thanks. Levy.-
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