华东某省的VoIP不能长期用！—— 说几分钟就。。。。cut左！！！
因为最近因为电话问题要从内网IP转换成公网IP，发现VoIP不能用！只说几分钟就cut左。非常明显，单不用5060 port，改用别的port number，移动系统都能侦测出来，我是用VoIP电话。所以下一步看看是否可以通过vmess到处理。 my situation similar
starting in Nov2018, all sip phones in Guangdong can register to hk elastix
and hear the voice but sip phones registered to same elastix cannot hear their voice!
change port, change public ip of elastix and those sip phones did not work...!
it seem sip connection blocked!
however, voice through iax is ok but quality is not that good as a phone call pass thro 2 elastix servers now.
Ching, any idea, i dont see any solution for your vmess on Google! I thought only me experienced this problem. Anyway, we have to do something which can hide our signalling and voice traffic across network broader. Since I was busy with using vmess (one of V2Ray protocols) to surf HK websites, I guess we can use similar method to get ride of this annoyed voip blocking problem. I need to speed some time to explore this possibility. Besides vmess, we may use other methods, for instance ss-tunneling method.
Your IAX method is a short-termed solution, we need a more secure method to overcome this issue.
As per your message, are you using two Elastix boxes, one in Guangzhou and the other in HK. Both of them are linked using the IAX protocol right? [i=s] 本帖最後由 角色 於 2019-2-9 11:21 編輯 [/i]
估计电讯公司用了Deep Packet Inspection （DPI）【1】
[quote]Unencrypted VOIP packets can very easily be detected. Skype is detected based on the initial connection and communication between two peers based on that initial connection is then disturbed or blocked.
Pure encrypted tunnels like VPNs carrying voice or Skype traffic cannot be detected by DPI devices.[/quote]
所以打电话需要secure calling【2】，但是这个secure calling是否都会让电讯商知道你在用VoIP呢？
References：
【1】[url]https://www.quora.com/Deep-Packet-Inspection-How-do-mobile-carriers-detect-VoIP-streams-inside-an-encrypted-connection-and-are-able-to-interpret-just-them[/url]
【2】[url]https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial[/url] Finally I got a solution for this issue. The initial problem came from desktop Siemens DECT IP phone. There is no encryption when connected to the HK Asterisk server. The traffic goes through the local ISP. I thought local IPS has done something in blocking VoIP traffic.
In this morning, I use mobile phone with the built-in data plan to make phone calls to one of my friends in Hong Kong and found the same problem (phone dropped after a few minutes talk). Based on my past experience, I have to run the VoIP over a VPN/proxy tunnels. Then I closed the VoiP app and started BifrostV to connect the remote HK V2Ray node to form a secured tunnel. I switched on VoIP app again. I can able to talk with my friend for more than an hour without any interruption.
After the call, I checked with my BifrostV data flow information. I indicated my VoiP app make phone call via proxy (the V2Ray server in HK) via the port 443.
For the case of the second post, you have to established a link built by V2Ray between your Guangzhou and Hong Kong office. Connect your phone call via the port 443 through the umbrella of V2Ray tunnel. There are other information which we may consider including:
[url]https://www.voip-info.org/sip-connection-based-on-ssh-tunnel/[/url] 廣東某地現在都ok, 只係講耐d就有好大雜音, 某一方聽唔到. 掛機重撥就ok [quote]廣東某地現在都ok, 只係講耐d就有好大雜音, 某一方聽唔到. 掛機重撥就ok
[size=2][color=#999999]99BB 發表於 2019-2-16 11:02[/color] [url=http://www.telecom-cafe.com/forum/redirect.php?goto=findpost&pid=46250&ptid=7340][img]http://www.telecom-cafe.com/forum/images/common/back.gif[/img][/url][/size][/quote]
exactly the same issue for me!
they detect the SIP signal and block you or mess up/delay your packet!!!
reboot the router cause a lot of issues in domain (ddns) update [b]回復 [url=http://www.telecom-cafe.com/forum/redirect.php?goto=findpost&pid=46171&ptid=7340]3#[/url] [i]角色[/i] [/b]
sorry for late return
yes
2 elastix box [quote]Finally I got a solution for this issue. The initial problem came from desktop Siemens DECT IP phone ...
[size=2][color=#999999]角色 發表於 2019-2-10 15:07[/color] [url=http://www.telecom-cafe.com/forum/redirect.php?goto=findpost&pid=46179&ptid=7340][img]http://www.telecom-cafe.com/forum/images/common/back.gif[/img][/url][/size][/quote]
I try to digest it
but it is a bit technical for me
i think i have to read and experiment a lot before I come back to you and report!
hope i figure out what to do
a thousand thanks for your advice
it is very kind of you to share your findings!!!! [b]回復 [url=http://www.telecom-cafe.com/forum/redirect.php?goto=findpost&pid=46180&ptid=7340]6#[/url] [i]角色[/i] [/b]
tks but this not work for me
as i wholly use cisco 7912 or cisco 7960 sip phone
and not implement voip under windows [i=s] 本帖最後由 角色 於 2019-3-9 18:11 編輯 [/i]
Instead of using SIP protocol, is there any IAX protocol to bridge two Asterisk boxes?
SIP --> Elastix -- via IAX protocol (through vmess using Dokodemo protocol)----Elastix ---> SIP [quote]Instead of using SIP protocol, is there any IAX protocol to bridge two Asterisk boxes?
SIP --> Elas ...
[size=2][color=#999999]角色 發表於 2019-3-9 17:58[/color] [url=http://www.telecom-cafe.com/forum/redirect.php?goto=findpost&pid=46404&ptid=7340][img]http://www.telecom-cafe.com/forum/images/common/back.gif[/img][/url][/size][/quote]
yes, it is our case use iax to bridge 2 asterisk
but i am still try to figure out how to use v2ray to build the tunnel [i=s] 本帖最後由 角色 於 2019-3-13 16:32 編輯 [/i]
The network configuration diagram may be as follows:
[attach]4318[/attach]
The iax signal from Asterisk 1 is sent to V2Ray node 1 (using the Dekodemo method) to connect the remote V2Ray node 2.
The v2ray can be easily installed using whatever method that you want. Should you have any problems, please let me know. [b]回復 [url=http://www.telecom-cafe.com/forum/redirect.php?goto=findpost&pid=46416&ptid=7340]13#[/url] [i]tsm[/i] [/b]
Based on my findings, using V2Ray's dokodemo (port forwarding method) should help you in resolving the VoIP issue when when you switch to use the single-port aix protocol [1,2].
In case you need to establish site-to-site, you may consider to use the OpenVPN method [1].
References:
[1] [url]http://www.telecom-cafe.com/forum/viewthread.php?tid=7394&extra=page%3D1[/url]
[2] [url]http://www.telecom-cafe.com/forum/viewthread.php?tid=7390&extra=page%3D1[/url] [b]回復 [url=http://www.telecom-cafe.com/forum/redirect.php?goto=findpost&pid=46416&ptid=7340]13#[/url] [i]tsm[/i] [/b]
In the last report, I mentioned that the packet-drop rate was extremely high. As a result, it would not be suitable for VoIP connection. However I found the cause of high drop rate was due to improper settings on my network. Now the problem has resolved and the OpenVPN connection can carry out ping packets with extremely low drop rate.
In this morning, I carried out an experiment using ping.[code]
sent=1640 received=1640 packet-loss=0% min-rtt=358ms avg-rtt=360ms
max-rtt=761ms
[/code]The reason for long ping time 360ms is China <--> US <--> Hong Kong. The packet travelling time is half of this figure, i.e. 180ms.
If you want to make it shorter, you can China <--> Hong Kong, the packet travel time will be reduced to 50ms.
If you have problem in setting up the V2Ray server, please let me know. SZ 也没办法使用了，BLOCK，切换到P S TN LINE都用不了，，SIP服务器都登录不上了Register Failed: No Response From Server。 [i=s] 本帖最後由 inshenzhen 於 2019-3-25 11:23 編輯 [/i]
ASTERISK 一样连不上 OBI成了没用的了 从上周开始发生的 VOIP无法正常登录了LINE拨出的电话 开始提示 拨打的号码是空号 你的Asterisk server放在香港吗？你的configuration是怎样？能多说一点吗? raspberry上建立在本地的 VPS国外的 两个互联。 raspberry上的能连OBI，但是LINE拨不出电话了，去掉OBI就没问题 [i=s] 本帖最後由 角色 於 2019-3-25 12:06 編輯 [/i]
都是看不懂！OBi我有，Asterisk servers我也有，你的LINE我知道是Obi里的LINE，接Public PSTN？那么你接哪里的PSTN？大陆？香港？建议手画一个network diagram for illustration。
你的VoIP packets不能通过普通的通道到另外一边server，一定要在V2Ray or alike + VPN这样的方式进行。 [quote]回復 tsm
Based on my findings, using V2Ray's dokodemo (port forwarding method) should help you in ...
[size=2][color=#999999]角色 發表於 2019-3-19 13:13[/color] [url=http://www.telecom-cafe.com/forum/redirect.php?goto=findpost&pid=46441&ptid=7340][img]http://www.telecom-cafe.com/forum/images/common/back.gif[/img][/url][/size][/quote]
Let me clarify your statement
if the connection between 2 elastix is IAX (using single port forward)
I can use V2Ray's dokodemo alone to connect 2 elastix and calls
if i use sip phone on 1 site to connect to elastix on the remote site
because sip connection require 5606 for handshaking and voice signaling thro port 10000 to 20000
I have to setup vpn (openvpn) to establish the connection and
accommodate multiple port forwarding for the voice call
in that case it should make reference to your V2ray dokodemo and openvpn thro it [b]回復 [url=http://www.telecom-cafe.com/forum/redirect.php?goto=findpost&pid=46491&ptid=7340]16#[/url] [i]角色[/i] [/b]
Thank you for your info and update
finally i see the light
[very depress and annoying in the past 3-4 months everyday getting jam for inhouse internal calls]
i will try to buy raspberry pi and see if I can ok. I agree it may be the most easy way without alternating a lot on existing network
[I am afraid to alter the router (shibby tomato on rt-n16) ]
I hope i can return with good news.
and there are so many articles here from u I have to spend time to digest and trial and error.
Tks again Thank you for your information about the choices of DDNS services. You are right that using mainland China's DDNS may cause problems as I experienced it before. The mainland Chinese DDNS does not resolve non-mainland-China IP.
As for your case, please do not worry as I shall create another post for your issue such that we can concentrate on the problems you have. [b]回復 [url=http://www.telecom-cafe.com/forum/redirect.php?goto=findpost&pid=46512&ptid=7340]22#[/url] [i]tsm[/i] [/b]
[quote]Let me clarify your statement
if the connection between 2 elastix is IAX (using single port forward)
I can use V2Ray's dokodemo alone to connect 2 elastix and calls
[/quote]
China-to-HongKong should be no problems. If you have public IP in China office, HongKong-to-China VoIP will not have problems. If you do not have public IP, you need to establish the reverse-proxy-method, which I have never tried it before.
[quote]
if i use sip phone on 1 site to connect to elastix on the remote site
because sip connection require 5606 for handshaking and voice signaling thro port 10000 to 20000
I have to setup vpn (openvpn) to establish the connection and
accommodate multiple port forwarding for the voice call
in that case it should make reference to your V2ray dokodemo and openvpn thro it
[/quote]
The answer is yes. [b]回復 [url=http://www.telecom-cafe.com/forum/redirect.php?goto=findpost&pid=46501&ptid=7340]21#[/url] [i]角色[/i] [/b]
谢谢你的解答，用的是大陆PSTN。另你的的帖子中里面的图和我连接的基本一样，今天我加了dokodemo转发好像是解决了，刚登录通话测试没问题。 [i=s] 本帖最後由 角色 於 2019-3-26 15:57 編輯 [/i]
能帮到你就好！
你有用Asterisk？Elastix？还有你用AIX？而不是SIP吗？因为其他member都会有你所类似的问题，能否说过一些吗？
如果再不用dokodemo，是否又不能工作？
然后又再用就OK？ 用的是FREEPBX 两个SERVER通过SIP连起来的，因为其中一个SERVER中的IAX有问题没有重装，就用了SIP。 你的意思，SIP那个port就通过dokodemo port forwarding就把问题解决好吗？
你大陆的router是否可以取得public IP？如果不是，你怎样从香港打上大陆的VoIP server呢？ YES I TESTED AT COMPANY
SURE IT IS PUBLIC IP
I WILL TEST COMPELETLY ON THE WEEKEND THEN REPORT HERE
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