similar to: RTP Packet Filterilng

In wiki pages it is stated that The audio channels (RTP) may go directly from phone to phone or may go through Asterisk's media bridge. Currently with my settings, I notice that all rtp's are passing through my asterisk. How could I achieve that they go directly from phone to phone? I assume this way, my machine will have less load and therefore could handle more calls. regards Bijan

Good day. I'm trying to configure termination with The Asterisk thru Cisco AS5300 Gateway from the SIP softphone (X-Ten X-Lite) to POTS network. I think, I had recognise kind of problem: call is ringing in the POTS phone (so I guess SIP signalling is working ok?), but there is no voice in either sides. On the Asterisk PC I can see incoming RTP streams with tcpdump and tethereal, but I

Hi, I'm connecting Asterisk v. 1.4.10 to Zanzibar Open IVR that acts as a SIP trunk. Since recognition didn't work correctly, I've troubleshot with Wireshark and saw that RTP stream is first send to one port on SIP trunk and then when first RTP packet arrives in opposite direction (from TTS part of Zanzibar - it's a prompt) Asterisk starts sending to the same RTP port -

Hello All When sending the Opus Packet in RTP Payload, the compressed frame is the output of the encoder? Also the config value as given in the RFC6716, 16...19 | CELT-only | NB | 2.5, 5, 10, 20 ms 16 corresponds to 2.5 ms 17 corresponds to 5 ms 18 corresponds to 10 ms 19 corresponds to 20 ms Is this correct representation of the data? Also in the RFC3551 the payload

Hi, Sorry if my question not related to speex. I have created a voip application using speex. My internet line is 64Kbps. I`m using narrow band mode, 8000 bitrate, 20ms of 8000Khz sampling rate, no buffering, no preprocessing, just set bitrate and go. When i send single frame inside rtp the sound was not good, choppy and noisy. But when i pack around 8 frame inside rtp packet, the sound was

I am working on a project that uses Asterisk ARI ExternalMedia request to stream the RTP audio from Asterisk to an UDP/RTP receiver project. Using slin16 format. 1) I believe I am seeing is a 12-byte header followed by 640 bytes of data. Is this correct? 2) Is there some place I can find a description of the 12-byte packet header fields? Dan -------------- next part -------------- An HTML

You cannot concatenate bytes because Speex frames don't necessarily end on octet boundaries. You need to call the encoder multiple times on the same SpeexBits bitpacket. Jean-Marc Manish Jalan wrote: > Hello, > > _*Background:*_ > The RFC 5574 suggests the RTP payload format for the speex codec. The > payload formation is straight forward; the encoded frames are to be >

I have tried to set up app_rxfax to receive faxes over IP. I realise there are mixed stories about how reliable this is at the best of times, but at this point all I'm after is some guidance in interpreting the log below. What does "RTP: Received packet with bad UDP checksum" suggest? Here is the full log: -- Executing SetVar("SIP/0892130888-b27c",

Hello Jean-Marc We really appreciate your input. If I understand it right, we should be calling the encoder on the same SpeexBits structure passing it a frame at a time to encode for as many times as the number of frames that we want to pack in the RTP payload. The output then obtained from from the encoder will have the necessary padding at the end without any separators between individual

I am trying to do an install of portupgrade. As root, I go to /usr/ports/sysutils/portupgrade and run make install distclean The system seems to obtain the pkgtools and some upgrades to it, but is unable to obtain the required bdb1-0.1.8.tar.gz. I have also looked at the collection at the freebsd site and tried the ftp site myself

I am looking for a method of dealing with the modified Bessel function K_\nu(x) for large \nu. The besselK function implementation of this allows for dealing with large values of x by allowing for exponential scaling, but there is no facility for dealing with large \nu. What would work for me would be an lbesselK function in the manner of lgamma which returned the log of K_\nu(x) for large

Hello, *Background:* The RFC 5574 suggests the RTP payload format for the speex codec. The payload formation is straight forward; the encoded frames are to be concatenated one after another. Once we have appended desired number of frames, we have to pad the stream with 01111 sort of sequence to ensure that payload ends on a octet boundary. *Observation:* I am using the speex encoder at 2150 Kbps

Hi, Am I correct in this.. I want to setup IPTABLES to protect my * box.. The default rtp.conf defines that * will use ports 10000 to 20000.. IAX listens on 5036.. SIP listens on 5060.. I am assuming all ports used by * are UDP.. So I am planning on setting my server to block all inbound traffic except UDP ports 5060, 5036 and 10000-20000.. Am I leaving anything out?? Thanks.. --

In the process of upgrading from 1.8 to 1.9 we are getting a lot of warnings about "Ruby 1.9 doesn''t support Unicode normalization yet". However the commit that added those lines is from 2008 and just mentions "Ruby 1.9 compat: no Unicode normalization support yet" without any references. Does anyone know whether this is still true for ruby 1.9 and for which minor

People are fond of free things, maybe more obviously during recession days. Not you? Well, you could still get through this article to know about how to create slideshow for free with slideshow making software. None of them would cost you a penny. Actually they probably exist in your computer system. So, just locate and run to start slideshow making, at least to kill boring time or relive

Revision: 65 Author: ajturner Date: Sun Dec 13 11:10:04 2009 Log: Use the GeoCommons addOverlay method http://code.google.com/p/mapstraction/source/detail?r=65 Modified: /trunk/source/mxn.geocommons.core.js /trunk/tests/index.htm ======================================= --- /trunk/source/mxn.geocommons.core.js Sun Dec 13 11:10:00 2009 +++ /trunk/source/mxn.geocommons.core.js Sun Dec 13

Hello! I created a program which injects a DLL into the game creation tool RPG Maker 2000/2003 (RPG Maker is written in Delphi 5/6) to add features. The DLL is injected by altering the code at the entry point. So there is some code run before the actual program starts. In this code, a WH_CALLWNDPROC hook is installed, and when RPG Maker opens its main window, some additional code will get

hi .. when placing a SIP call to a sip host in the states every few seconds I get an RTP codec 19 error. I know this is related to comfort noise, and the call goes through OK ... how can I suppress the error message ? Also, many times I get "Invalid CSeq Number" back from 216.52.153.207 (which is the host i'm calling) and the call drops.. is there a solution for this ? cheers Dave

the above-message keep popping up every second during a conversation between a zap(fxs) channel and sip channel. * eventually hung after a long while we can talk to each other and we can ring one another without any problem. i've had x-lite and x-pro register with * without this problem. furthermore, i have ask my friend to turn off all codec expect g.711MLAW; that did not help i then

After just having update to the latest CVS I am getting the following message when I call VoicemailMain(): -- Executing VoiceMailMain("SIP/2302-1a12", "") in new stack -- Playing 'vm-login' (language 'en') -- Playing 'vm-password' (language 'en') -- Playing 'vm-youhave' (language 'en') -- Playing