Question

我有时会用星号加密卡住。

sip.conf重新加载没有任何问题，作为拨号计划，注册SIP客户端 - 完全没问题

当我打电话给表格一个zoiper sip帐户到另一个wireshark捕获tcp eth流量显示以下行：

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 - 星号服务器

192.168.13.252 - android phone（zoiper）

在通话期间，两部手机都没有声音问题。两部手机都发送包裹但没有接收任何包裹。

这是涉及的SKYPE协议吗？它假设是RTP协议的所有行。

Answer 1

如果您通过SIP注册但没有收到音频，那么由于某种原因，用于RTP的较高端口很可能没有接收到数据。通常这些端口是10000-20000。确保两个IP可以通过端口5060-5061和更高端口相互通信。您是否可以在尝试拨打电话时显示星号CLI输出？

asterisk -vvvvvvvvvvvr

Answer 2

很高兴让我们了解一些细节。 rtp.conf

[general]
rtpstart=10000
rtpend=20000

没有错误重新加载sip。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，一切都井然有序。 Asterisk在本地IP上运行，没有防火墙。

asterisk sip.conf tls加密

2 个答案: