3
我已经创建了一个类似于STUN的+ rendez-vous服务器。 我已经尝试过WIFI(国内NAT后面)的一切,并且一切正常。 我有两个移动ISP,一个允许一切(VOIP + P2P +调制解调器)(ISP 1) 另一个允许除P2P(ISP 2)以外的一切。通过3G网络的VOIP
尝试通过ISP 1时,它也可以正常工作。 但是,当我尝试与ISP 2,udp数据包不通过。
我已将我的计算机连接到ISP 2上的电话,并运行TUM NAT分析器。
它告诉我
UPnP Test (?): No UPnP device found
STUN Test (?): Symmetric NAT
UDP Binding Test (?): Endpoint depenent binding, port prediction may be hard
TCP Binding Test: Endpoint depenent binding, port prediction may be hard
UDP Mapping Test (?): local and external IP addresses were different
(NAT). Your source ports were not preserved. It may be hard to predict your external source port.
TCP Mapping Test: local and external IP addresses were different (NAT).
Your source ports were not preserved. It may be hard to predict your external source port.
SIP ALG (?): The initial SIP INVITE packet has not been modified on its way to our servers.
There is no SIP ALG involved
FTP ALG: The initial FTP PORT command has been modified.
Most probably, your NAT implements a FTP-ALG
因此很明显,它使用随机端口做作(没有办法使用显然端口预测)对称NAT。
所以我想知道,允许VOIP但不是P2P(并且没有SIP ALG)的ISP,是否期望VOIP使用中继服务器来工作?
或者我错过了什么......? 据我了解AT & T(及可能其他人)使用相同类型的NAT作为我的ISP 2 ...(对称NAT),以便成为一个巨大的问题,我想....
任何不过,想法,反应会很好。