VoIP Blogging > Maxim Sobolev RTPproxy

[VOIP-info.org Wiki Changes] RTPProxy looks for an existing sessions with such id, if the session exists it returns UDP port for that session, if not, then it creates a new session, binds to a first empty UDP port from the range specified at the compile time and returns number of that port to a SIP Proxy. After receiving reply from the proxy, SIP Proxy replaces media ip:port in the SDP to point to the proxy and forwards the request as usual;

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[3CX VOIP, SIP & Phone System blog» - 3CX VoIP, SIP, Phone System blog] 3CX VOIP, SIP & Phone System blog » Connecting your Home Phone to 3CX: So, I decided to start simple and take it step by step.  I began by simply enabling STUN and NTP on the device and setting the registration and proxy servers to the external DNS name of my corporate 3CX system (note I already have UDP 5060 for SIP mapped into my PBX along with UDP 9000-9015 for audio).  I popped up the admin interface to 3CX…and….nothing.  No registration.  OK, no problem, try again.

[Musimi.dk forum] Cisco 7940, Airport Extreme, Yousee: SIP Default Generic Configuration File # Image Version image_version: "P0S3-8-12-00" # Proxy Server proxy1_address: "musimi.dk" ; Can be dotted IP or FQDN proxy2_address: "" ;

[TechNet Blogs] Tonino Filipovic : Office Communicator SIP Trace - Voice Call Flow: The reason behind involvement of Outbound Routing component is rather simple: E.164 number of the called party (+38511234567) could not be matched to any existing SIP URI, which shows that the called party is either on a PBX system or PSTN, so the call has to be routed out of OCS system to the next hop using Local Calls route (PhoneRoute="Local Calls";Gateway="R2MS01.uclab.org:5061"). Just a note that apart from finding the best route for the call, Outbound Routing also performs a call authorization (in a nutshell, it is matching normalized number and caller’s Voice Policy phone usages to the number pattern and phone usages in available routes).

[Binod's Blog] Locating SIP servers using DNS in SailFin | Java.net: Lets take the NAPTR record shown above, the third column, in these records suggest that the server support TLS over TCP, TCP and UDP protocols and are preferred in that order. sip-proxy.war : A sip application as explained in Sip Proxy example of sailfin.

[CCIE Routing & Switching Technical] Unblock VoIP in UAE/Oman/Qatar/Africa - IEOC - Internetwork ...: Border Controller(VGBC) at customer CPE side, VGBC can work in the way similar to that of soho router, but it only encrypts and decrypts SIP and RTP packets on link layer, not to handup these packets to IP stack for forwarding while bypassing other data packets originating from SIP terminals. In this scenario, peak throughput and minimal CPU overhead can be easily achieved.

[jfarcand's blog] Writing a TCP/UDP stack supporting the SIP protocol using the ...: In between the TCP stack and the sip server I need some kind of gateway to translate TCP messages to SIP and vice versa. I need to built that gateway/proxy kind.

[Twisted] Changeset 27464 - Twisted: return '<URL %s:%s@%s:%r/%s>' % (self.username, self.password, self.host, self.port, self.transport)

[Teodor Georgiev's technical blog] Teodor Georgiev's technical blog » Blog Archive » Quintum vs Cisco: It is free of charge (unlike the Cisco buggy and ugly HTTP interface, or expensive third-party products like the one of Telcony) and can be used even when the Quintum gateway is behind a NAT with a private IP (in this case the gateway is the one, who initiates the connection to the Tenor Configuration Manager).

[Voip Blog] Asterisk tip: P2P SIP URI Dialing | Voip Blog: SIP and IAX2 are also host names that point to my PBX. When someone dials my URI extension@blyon.com, their client or PBX will do a DNS lookup and see that sip is available on port 5060 at sip.blyon.com.

[Forefront TMG (ISA Server) Product Team Blog] Forefront TMG (ISA Server) Product Team Blog : Forefront TMG is ...: A SIP VoIP call is carried out using User Datagram Protocol (UDP), and incorporates two protocols: Session Initiation Protocol (SIP) for call establishment and termination, and Real Time Protocol (RTP) for media (audio and/or video). SIP can also be carried out using Transmission Control Protocol (TCP) but for the purpose of this post I will refer to SIP carried out using UDP.

[JimmyRay's blog] Overlooked VOIP Security Features | NetworkWorld.com Community: So I waited for the cleaning crew to arrive and I grabbed my red banged up tool box and heading in the front door with everyone else and just acted like I know where I am going and I belonged there. It helps to be walkin’ and talkin’ with someone as well because most folks will not want to interrupt your discussion.

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