sip proxy

* RFC 2327 (SDP: Session Description Protocol) * RFC 2782 (dns srv Resource Record) * RFC 3261 (SIP: Session Initiation Protocol) * RFC 3263 (locating sip servers) * RFC 3264 (an offer/Answer Model with SDP) * RFC 3265 (SIP-specific event notification) * RFC 3266 (support for IPv6 in SDP) * RFC 3403 (DNS naptr Resource Record) * RFC 3420 (Internet media t

Tags: CME cucm Cisco UnifiedThere is a logical association between the commands and must be configured in a sequential order.1. Configure voice encoding and SIP address formatVoice class URI 2 sipHost IPv4:address of CUCMVoice Class codec 1Codec Preference 1 G729R8Codec Preference 2 G711ulawCodec Preference 3 G711alawCodec Preference 4 g722-642. Configure VoIP Service Specific parametersVoice service VoIPIP Address Trusted ListIPv4 The address of the

Java Dynamic Proxy, proxy and InvocationHandler, Proxy dynamic proxy I have read a lot of articles about proxy, understanding and sorting out them. 1. Basic composition of proxy Abstract role: Declares the common interfaces of the

The SIP protocol and some of its applications are worth learning. In this regard, we will explain the configuration of rtp sip today. SIP (Session Initiation Protocol) is usually used for VOIP calls, call establishment, call negotiation, and call termination. it helps two terminals to recognize each other, but it does not process media. When A call is established

Processing of FreeSWITCH SIP signaling in Mod_sofiaFirst, a thread that handles SIP messages is created inside the module's load (mod_sofia_load): /* Start one message thread /switch_log_printf (Switch_channel_log, Switch_log_info, "starting initial message Thread.\n "); Sofia_msg_thread_start (0); The Config_sofia function is then called. if (Config_sofia (Sofia_config_load, NULL)!= switch

Label:Recently studied asterisk configuration, before the SIP account is configured in the sip.conf file, manual writing dead, the current demand, is the dynamic new SIP account, saved in the database.After adding data to the database, use the command SIP show users to not load the SIP account to the database.1. Downlo

The integration of unified communication involves software, hardware, mobile devices, and fixed devices. The standards and protocols are different. Therefore, the integration is difficult. In terms of IP phones and desktop applications, there are mainly SIP and H.323 Protocols; mobile videos are H.324M; Monitoring also includes H.323, SIP and proprietary protocols as well as MEPG2, MEPG4, and H. 264 and ot

Practice. It is not enough to know some knowledge. You need to practice it first. Now we have learned about the SIP protocol. Here we will share the practice process of a netizen's sip invite. I hope it will be useful to everyone. Request sent by linphone in sip invite (reguest) INVITEsip:to@192.168.105.14SIP/2.0 Via:SIP/2.0/UDP192.168.105.5:5060;rport;br

Research on ice-based sip signaling penetration over symmetric NAT technology Zeng Li, Wu Ping, Gao Wanlin, Wu wenjuan (Department of Computer Science and Technology, Agricultural University of China, Beijing 100083, China) 2 (School of information, Renmin University of China, Beijing 100872, China) Abstract what is one of the practical difficulties faced by IP-based speech, Data, video, and other services in the NGN network?Effectively Penetrate vari

In my previous article, I introduced how to use a SIP Soft Phone for direct calls. However, if many other users need to talk to each other, at the same time, authentication and control of User Account logon is required based on security considerations, in these cases, you need to establish a sipserver.Role of sipserver:Call control and processing functions, business provision/support functions, user management functions, protocol processing functions,

I am responsible for custom development of the SIP/IMS video client and support access to the SIP Soft Interface.Switch, IMS core network, supportedVoice, video, and instant messaging functions. The video formats support h263, h264, and MPEG4 soft encoding solutions. The hardware coding/decoding interface is provided for interconnection and servers. If you are interested, contact me. Csdn lidp http://blog.

Haha. I 've talked about some SIP application scenarios. I will write some theoretical and boring things in the future. Before that, let's continue to make it easy to tell you a story about how to learn HTTP. At the very beginning, I took an HTTP book first. In short, it was a brick book. After reading the two chapters, we found that we should use the knowledge of TCP/IP. So let's look at TCP network programming. Two months later, I still don't kno

After OS X upgrades to El Capitan, it provides a security-related pattern called SIP (System Integrity Protection), also known as rootless mode, which is a new feature that emphasizes security for OS X, which prohibits the software from being used as root in Mac running on, upgrade to OS X 10.11 Maybe you'll see some apps are disabled so that/usr/bin folders we can't read and write properly, but it also causes some programs (such as homebrew and Git)

Developed for SIP/IMS video clients, supports access to sip Softswitch, IMS core network, andVoice, video, and instant messaging functions. The video formats support h263, h264, and MPEG4 soft encoding solutions. The hardware coding/decoding interface is provided for interconnection and servers. If you are interested, contact me. Implements standard-based (SIP

In the process of building PYQT I met a very disgusting problem, in the installation of SIP after compiling the source of the installation process has been prompted me: Operation not permitted , I even reinstall the system is useless, and finally through the data to solve the problem. Installing SIPDownload the SIP source package after extracting it into its directory:python configure.pysudo makesudo m

Release date:Updated on: Affected Systems:Yealink Yealink SIP-T20P IP Phone Description:--------------------------------------------------------------------------------Bugtraq id: 57029Yealink SIP-T20P is an IP Phone.YeaLink IP Phone SIP-TxxP The vulnerability is described as follows:1) The default username ("user") and password ("user") can access the hidden pa

1 Description This article describes in detail the P2P SIP telephone process based on the STUN series protocol, which involves the interaction of SIP signaling, the principles of P2P, and Protocol interaction of STUN, TURN, and ICE. The interaction between service units mentioned in this article uses UDP, which does not involve TCP holes and other TCP-related operations. This document assumes that neither p

Both SIP and XMPP are application-layer protocols that are used primarily to send voice and instant messaging over the Internet im,rfc3521 defines the sip,rfc3920 definition of XMPP. XMPP comes from instant messaging systems, SIP-like voice and video communications.XMPP protocol is mainly responsible for the exchange of data,

Is the call flowchart of Asterisk: We use the call process of SIP as an example to describe the call process of other channels. The call process (incoming) is as follows: Do_monitor-> sipsock_read-> handle_request-> handle_request_invite-> sip_new/ast_pbx_start-> pbx_thread->__ ast_pbx_run -> Ast_spawn_extension-> pbx_extension_helper-> pbx_exec-> execute dialplan When the chan_sip module is loaded, an independent listening thread do_monitor is starte

Haha, if you have never touched network programming, don't look down. Give a definition first: SIP (Session Initial Protocol) is a signaling protocol that is used to set up, modify and terminate sessions, like Internet phone calland multimedia conferences between two participant ants. For Translation: SIP is a signaling protocol used to establish, modify, and terminate sessions, such as network call