Tags > VOIPtechnologytoinvestigate
- 03-24otherTop media is freeswitch, after kurento docking through kurento video content are vague, strives for
- 01-27otherSIP cancel the timer process
- 01-27otherAn asterisk and freeiris2 relations
- 12-07otherHuawei huawei p20, P20PRO account number and password all don't remember what lost account to p
- 12-02otherQq friends each other had deleted me how can I restore friends add
- 11-30otherFor a great god!!!!! Ali cloud centos7 system installation asterisk1.8.7.1 started after 5060 port i
- 11-30otherFreeSWITCH no call in the console
- 11-28otherSip with the customer after the docking, customers can dial the telephone, but could not hear the Lo
- 11-24otherVOIP: VOS3000 operation instructions to download
- 11-23otherESXI unable to login
- 11-18otherAt one end is a symmetric NAT, can realize p2p make hole
- 11-07otherThe sip protocol in the dial-up process without ringing message, excuse me, is what problem?
- 11-04otherVoip + 3 cx how to realize network telephone communications
- 11-04otherSocket network communication programming, there is no link between two network card to connect direc
- 11-04otherWhat thing
- 11-04otherOpensips server netcom - failure
- 10-30otherCisco 7945 g can't registration, online, etc., be urgent!!!!!!
- 10-22otherAndroid simple calculator source code...
- 10-20otherHow to set up public SIP server, such as FreeSwitch, miniSipServer
- 10-18otherAn asterisk configuration IVR remove user input user name
- 10-18otherWeb page version of the telephone communications development based on sip protocol
- 10-16otherMulticast performance degradation, when switches have great god know to ask?
- 10-16otherImplement a UAC eXosip, can only send can't accept it
- 10-14otherCompile smallmgc project finally appeared a lot of "undefined"
- 10-13otherPlease ask a question of IP
- 10-11otherFor help! FXO gateway ran into problems
- 10-09otherSIP Servlet development, Tomcat is how to identify the SIP. The XML, can directly add the SIP in web
- 10-09otherExosip as the service side, after receiving the SIP messages, if access to the source in the udp por
- 10-09otherUnder WINDOWS install cygwin XWin last with ns2 sever run wrong
- 10-08otherHave a great spirit about h323plus information compiled under the MAC
- 10-08otherVOIP + IM skype open source system
- 10-08otherBased on the sip protocol and h. 323 protocol Internet gateway technology
- 10-07otherWhich master can give me a more detailed explanation of RTP/RTCP SSRC and the use of a CNAME
- 10-06otherNovice first asterisk, how do I quick start
- 10-06otherAn asterisk how to realize the calling terminal
- 10-06otherSmall white, see the wireshark packets of data and send them by imitation of PHP custom data
- 10-05otherAfter centos7 access networks, with a segment of the network packet loss
- 10-05otherExcuse me freeiris problem
- 10-05otherAbout voip telephone
- 10-05otherVideo with TCP, audio RTP, spread, how to make audio and Video synchronization?
