Asterisk支持push to talk功能吗？

langfanyun

如题

ziggler

http://blog.csdn.net/lin_bei/archive/2008/05/31/2497750.aspx

好像不支持。

langfanyun

有没有其他开源的或商业的Sip server支持push to talk 功能的了？

ziggler

*
      Open Source SIP Libs


Name         Describe        
Asterisk         Asterisk is a complete PBX in software. It runs on Linux, BSD and MacOSX and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in many protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.        
Minisip         Minisip is a SIP User Agent ("Internet telephone").
It can be used to make phone calls, instant message and videocalls to your buddies connected to the same SIP network.        
oSIP stack (GNU)         oSIP implements the Session Initiation Protocol (published by IETF as RFC 3261). It can provide signalling capabilities for multimedia applications (IP phones, etc.). It provides a fully usable parser for the SIP syntax and implements the "transaction layer" as defined in the draft. It also provides an SDP parser and extra features for the User Agent. It can be used to build both proxy and IP phones.        
*pjsip
*oldsite         Small footprint SIP stack         
Partysip         SIP proxy server        
SIPfoundry.org                
SIP Express Router (SER) (iptel.org)         SIP Express Router (SER) (iptel.org)
ptel.org SIP Express Router" (SER) is a high performance configurable, free server implementing Session Initiation Protocol (SIP).

It can act as registrar, proxy server, redirect server and server many roles in SIP networks, including PSTN gateway guard, SMS/Jabber gateway and application server.

http://www.iptel.org/ser/

iptel.org
Description: iptel.org collects related IETF documents and links to other resources related to SIP. The site offers free SIP accounts. SIP Express Router, free SIP server, is available from iptel.org as well.
       
sipsak        

sipsak is a small command line tool for developers and administrators of Session Initiation Protocol (SIP) applications. It can be used for some simple tests on SIP applications and devices. sipsak is a "swiss army knife" for SIP developers.

sipsak is an open source tool available at http://sipp.sourceforge.net/. The home page for the tool is at http://www.sipsak.org/.
       
SUN         Sun also has a number of open source SIP projects:

   1. A Java SIP stack compliant to JSR32 (http://jain-sip.dev.java.net)
   2. A proxy server application and a presence/IM user agent that runs on JSR32 (http://jain-sip-presence-proxy.dev.java.net)
   3. A soft phone with IM capabilities that runs on JSR32 (http://sip-communicator.dev.java.net)

       
Vovida         a communications community site dedicated to providing a forum for open source software used in datacom and telecom environments        
Yate - Yet Another Telephony Engine        

ATE is a telephony engine designed to implement PBX and IVR solutions for small to large scale projects.

Yate can be used as a:

    * VoIP server
    * VoIP client
    * VoIP to PSTN gateway
    * PC2Phone and Phone2PC gateway
    * H.323 gatekeeper
    * H.323 multiple endpoint server
    * H.323<->SIP Proxy
    * SIP session border controller
    * SIP router
    * SIP registration server
    * IAX server and/or client
    * IP Telephony server and/or client
    * Call center server
    * IVR engine
    * Prepaid and/or postpaid cards system

       
sofia-sip         Sofia-SIP - a RFC3261 compliant SIP User-Agent library.        

SIP Resources - protocols - live as garfield (26 May 2009)

http://blog.chinaunix.net/u/17978/showart_125912.html

langfanyun

ziggler兄，
能告诉我哪一个server支持ptt了？