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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
Server Stats
  • Execution time: 0.37s
  • Memory usage: 2.21MB
  • Database queries: 32
  • GZIP: Disabled
  • Server load: 2.77

RTP

RTP opens two ports for communication. One for the media stream (an even port number) and one for control (QoS feedback and media control) - RTCP. The port numbers are not hard defined, it depends very much upon the application.

  • RTP (Real-time Transport Protocol)
  • RTCP (Real-time Control Protocol)
    • Adds information for:
    • Packet Loss
    • Jitter
    • Delay
    • Signal Level
    • Call Quality Metrics
    • Echo Return Loss
    • etc.
  • RTCP XR (Real-time Control Protocol Extended Reports)
    • All of the RTCP list above plus:
    • R Factor
    • MOS
    • and more

Actual voice packets are sent using RTP/RTCP for SIP VOIP calls. RTP is able to carry media identified by parameters registred by the Internet assigned numbers authority, IANA. These are also used for SDP descriptions in SIP and MGCP messages.

Some of these payloads:



PT encoding name audio/video (A/V) clock rate (Hz) channelsRef
0 PCMU A 8000 1 RFC3551
3 GSM A 8000 1 RFC3551
4 G723 A 8000 1 Kumar
5 DVI4 A 8000 1 RFC3551
6 DVI4 A 16000 1 RFC3551
7 LPC A 8000 1 RFC3551
8 PCMA A 8000 1 RFC3551
9 G722 A 8000 1 RFC3551
10 L16 A 44100 2 RFC3551
11 L16 A 44100 1 RFC3551
12 QCELP A 8000 1 -
13 CN A 8000 1 RFC3389
14 MPA A 90000 RFC3551,RFC2250
15 G728 A 8000 1 RFC3551
16 DVI4 A 11025 1 DiPol
17 DVI4 A 22050 1 DiPol
18 G729 A 8000 1
19reservedA
20unassignedA
21unassignedA
22unassignedA
23unassignedA
24unassignedV
25CelBV90000 RFC2029
26JPEGV90000 RFC2435
27unassignedV
28nvV90000 RFC3551
29unassignedV
30unassignedV
31H261V90000 RFC2032
32MPVV90000 RFC2250
33MP2TAV90000 RFC2250
34H263V90000 Zhu
35--71unassigned?
72--76reserved for RTCP conflict avoidance RFC3550
77--95unassigned?
96--127dynamic? RFC3551



RTP and NAT

In a VOIP session, there are two RTP streams, one in each direction. If one of the parties involved in the session is on a private IP address, that stream from the public client to the NAT box, will not be allowed to reach the client on the inside of the NAT. To handle this, Symmetric RTP is often used. For more information on NAT and VOIP, see NAT and VOIP.

Articles


RFCs

  • IETF RFC 3550 RTP: A Transport Protocol for Real-Time Applications
  • IETF RFC 3611 RTP Control Protocol Extended Reports (RTCP XR)
  • IETF RFC 1890 RTP Profile for Audio and Video Conferences with Minimal Control
  • IETF RFC 2508 Compressing IP/UDP/RTP Headers for Low-Speed Serial Links
  • IETF RFC 3545 Enhanced Compressed RTP (CRTP) for Links with High Delay, Packet Loss and Reordering


See also


Created by jht2, Last modification by admin on Sun 11 of Mar, 2007 [06:12 UTC]

Comments Filter

RTP Server Component?

by Matthew on Friday 26 of May, 2006 [16:38:11 UTC]
What is the actual RTP server component(s) used by Asterisk? How do I replace the default RTP server in the standard Asterisk distribution with another one, whether I downloaded it or wrote it myself? Where in CVS and descriptions of interfaces (APIs, network) can I look for the boundaries of what the new RTP server must support?

Re: voice quality

by admin on Thursday 20 of April, 2006 [15:57:01 UTC]
I believe that like G.729 Codecs, there are patent/licence issues that prevent a free implementation of MOS, PSQM, etc.

voice quality

by Pawel on Tuesday 18 of April, 2006 [19:44:58 UTC]
Hi !
Does anyone know free software, to measure voice quality in MOS scale (P.800, PSQM, or whatever)? I spent a lot of time on google but didn't find anything free :(

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