asterisk sip.conf tls加密

Question

我有时被星号加密卡住了。

sip.conf重新加载没有任何问题，作为拨号方案，注册sip客户端-没有任何问题

当我从一个zoiper sip帐户呼叫另一个wireshark capture tcp eth流量时，显示以下行：

192.168.13.252    192.168.13.253    RTP    224    PT=ITU-T G.711 PCMU, SSRC=0x4C8C7A63, Seq=2259, Time=3154311440
192.168.13.253    192.168.13.252    SKYPE    224    Audio Unk: 5
192.168.13.253    192.168.13.252    SKYPE    224    Audio Unk: 5
192.168.13.253    192.168.13.252    SKYPE    224    Audio Unk: 5
192.168.13.252    192.168.13.253    RTP    224    PT=ITU-T G.711 PCMU, SSRC=0x4C8C7A63, Seq=2260, Time=3154311600
192.168.13.252    192.168.13.253    RTP    224    PT=ITU-T G.711 PCMU, SSRC=0x4C8C7A63, Seq=2261, Time=3154311760
192.168.13.253    192.168.13.252    SKYPE    224    Audio Unk: 5 ...

192.168.13.253 - asterisk服务器

192.168.13.252 - android手机(zoiper)

问题是打电话时两部手机都没有声音。两部手机都会发送包裹，但不会接收任何包裹。

那是不是涉及SKYPE协议呢？它应该是RTP协议所有行。

Answer 1

如果您通过SIP注册，但没有收到音频，则很可能是由于某些原因，您用于RTP的较高端口无法接收数据。通常这些端口是10000-20000。确保两个IP可以通过端口5060-5061和更高的端口进行通信。您是否可以在尝试进行呼叫时显示星号CLI输出？

asterisk -vvvvvvvvvvvr

Answer 2

太好了，让我们来了解一些细节。rtp.conf

[general]
rtpstart=10000
rtpend=20000

重新加载sip时没有错误。这很有趣：

####CLI ### asterisk -vvvvvvvvvvvr #### shows
 == Using SIP RTP CoS mark 5
       > 0x7fb264004c00 -- Strict RTP learning after remote address set to: 192.168.13.104:58136
    -- Executing [200@phones:1] Dial("SIP/201-0000000b", "SIP/200") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/200
    -- SIP/200-0000000c is ringing
       > 0x7fb2440062f0 -- Strict RTP learning after remote address set to: 192.168.13.106:62856
    -- SIP/200-0000000c answered SIP/201-0000000b
    -- Channel SIP/200-0000000c joined 'simple_bridge' basic-bridge <9726e2bc-f161-452c-b489-c1829af2ed70>
    -- Channel SIP/201-0000000b joined 'simple_bridge' basic-bridge <9726e2bc-f161-452c-b489-c1829af2ed70>
       > 0x7fb264004c00 -- Strict RTP switching to RTP target address 192.168.13.104:58136 as source
       > 0x7fb2440062f0 -- Strict RTP switching to RTP target address 192.168.13.106:62856 as source
       > 0x7fb264004c00 -- Strict RTP learning complete - Locking on source address 192.168.13.104:58136
       > 0x7fb2440062f0 -- Strict RTP learning complete - Locking on source address 192.168.13.106:62856
    -- Channel SIP/201-0000000b left 'simple_bridge' basic-bridge <9726e2bc-f161-452c-b489-c1829af2ed70>
    -- Channel SIP/200-0000000c left 'simple_bridge' basic-bridge <9726e2bc-f161-452c-b489-c1829af2ed70>
  == Spawn extension (phones, 200, 1) exited non-zero on 'SIP/201-0000000b'
####

根据CLI控制台信息，一切正常。Asterisk在本地IP上运行，无防火墙。