similar to: Can Asterisk send RTP to a specific port number?

(sent on behalf of Federico Montesino Pouzols <fedemp@altern.org>) Hi, I like the draft, particularly because of its simplicity, which is another pro of speex compared with other codecs. Some comments: <p> - Reading section 3, I understand that frames cannot be fragmented across different RTP packets, which seems quite reasonable. I think a explicit statement on this

When a SIP call starts (INVITE / 200 OK), asterisk seems to create a random port number for voice (rtp) packets. Is it possible to force this port value (without using reinvite since i am trying to use SIP against something else than sip) thanks a lot in advance -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have an Asterisk 11.5 system, using SIP Realtime and operating as a ITSP. One of my customer's endpoints is a NetVanta 7100 PBX system that has a SIP trunk connection to my Asterisk box. The NV 7100 has a public IP on it that doesn't have any NAT between it and my Asterisk system. When the customer transfers a call from one handset to a voicemail box, the NV 7100 sends a RE-INVITE to

Hello, I have defined that I want to receive audio (RTP) on port 11500 till 11954 (rtp.conf). The same range I have defined in my firewall. I now see that an IP-address gets blocked by my firewall because there are packets coming onto port 11955. How come the client sends audio on port 11955 when I clearly define in my SDP-body that I want to receive audio on port range 11500 till 11954 ?

Hello: I have this situation: I can make calls internally, I can make inbound calls but I can't make outbound calls. Thanks in advance. These are my devices: * asterisk 11.8.1 = 192.168.1.22 * sipphone grandstream gxp2160 = 192.168.1.5 * gateway audiocodes mp-114 2fxs-2fxo = 192.168.1.4 port 1 (FXS) connected to an analog phone port 3 (FXO) connected to the PSTN These are my

The latest revision of the draft RTP Profile is attached for review. This will be submitted to the IETF Audio-Video Transport Working Group for consideration immediately, so if you have any more comments, let us know. In addition, we will be applying for an official MIME type. Note that the AVP code and the MIME type in this latest revision have been changed from "SPX" to

I have 2 sip accounts setup - 200 and 202. If I do sip show peers I get: sip show peers Name/username Host Dyn Nat ACL Mask Port Status 202/202 192.168.0.6 D 255.255.255.255 5060 Unmonitored 201/201 (Unspecified) D 255.255.255.255 5060 Unmonitored 200/200 192.168.0.3 D 255.255.255.255 5060

if the new draft in the manual is used. I don't find how to tell the decoder which mode(NB/WB/UWB) is used in the encoder. The RTP header don't contain the mode field and I don't find the mode information in the coded frame either. Does this mean we have to use NB decoder in all cases? Lianghu On 11/22/06, Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > >

Hello, short question : does Asterisk reserve RTP ports for every IP-phone that is being called ? If for instance an incoming call makes 10 IP-phones ring, does this mean that Asterisk preserves 10 x 2 RTP ports for audio ? I guess Asterisk sends in the SIP INVITE an SDP body with an RTP port number for audio ? If this is the case for the 10 IP-phones to which an INVITE is send to, this

Dear Opus developers. This is the first time I write here, so hello to everybody! Sorry to disturb you but I would like to ask you something I could not answer by googling and by looking at this mailing list archive. I have just started investigating this new and promising codec for real-time audio transport over the internet for industrial applications. I was previously experimenting with RTP

On Fri, Sep 1, 2017 at 9:13 AM, Joshua Colp <jcolp at digium.com> wrote: > On Fri, Sep 1, 2017, at 09:01 AM, Dave Topping wrote: > > http:/www.theregister.co.uk/2017/09/01/asterisk_admin_patch/ > > This specific issue exists in a lot of different implementations and > devices. Unfortunately there's nothing within SDP that guarantees or > provides what the source of

Hi all We are about to send an updated version of the internet draft "RTP Payload Format for the Speex Codec" to the IETF AVT working group. Before submitting we would like your input, if you have any comments or input please send them to the mailing list. If we don't get any comments in 1 week (by 22. May 2007) we will go ahead and submit it to the IETF. Of course you can comment

I have spent the morning on this project, still without success. Summary: Yesterday I inadvertently unplugged my Grandstream phone. I might add I did a rebuild of my s/w from CVS at the same time. Since then, the Budgetone seems to talk SIP just fine, but the RTP being sent to it by asterisk "doesn't make any sound." It was suggested I do a factory reset of the phone, which I

Hi - I am currently developing a RTSP/RTP/SDP solution to stream Speex encoded data. Using my current source, I have successfully streamed u-law and PCM encoded audio but have been unsuccessful thus far with Speex. Because of some constraints of my system, I am encoding audio at 11.025kHz. I am still using the 160 samples per frame which makes my frame size 28 bytes. I have successfully

Dear All, I am having some problems with connecting with a UA. Sometimes there is not sound in the call made, sometimes the caller would near no sound, while the callee can hear the caller. I have attached the rtp debug and sip debug for you comments. Please help me. Thank you all. Asterisk Version is 1.2.1 Asterisk RTP Range is 10000 to 20000 UA Listen RTP Port is 15000 Below is the the

Been working with Asterisk for a long time but this is the first time I have dealt with this issue. I am setting up an Asterisk box (FreePBX not my choice) to interface with an e911 provider. They say their switches only listen for RTP on ports 20000-21001 which is outside the normal range Asterisk listens on 10000-20000. I wish I knew more about this topic but since I have never had an issue

On 2/23/19 4:19 PM, Joshua C. Colp wrote: > On Sat, Feb 23, 2019, at 11:04 AM, hw wrote: > > <snip> > >> >> directmedia is not explicitly enabled; I guess it's the default. >> >> Joshua basically says there is no way to control which ports are being >> used for SRTP because that it is "up the endpoint". Such endpoints, in >>

hi, i have following topology PSTN - Asterisk ---- internet -----  router - jssip client (wss) Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp for SIP connection to PSTN router - public IP/private IP (NAT) jssip client - private IP - sip over websocket to Asterisk PJSIP ~30% of calls has problem with no audio. reason is that Asterisk is sending RTP to private IP of jssip

Hello List, I need your advise please. I am trying to establish a call from Asterisk 1.8.15-cert5 to one remote SIP UA (not Asterisk), both are behind NAT. That remote peer is configured with nat=yes in my sip.conf but yet RTP packets are being sent to its internal IP address which is declared in the Connection Information (c) in the SDP, obviously reaching nowhere. I need RTP to be sent to the

Hi all, I've an asterisk 1.2.5 running very well for about a 9 months, and suddenly i cannot dial extensions 4XXX from SIP Phones. Now comes the wired stuff... I can dial this extensions from IAX phones as well as from Analogue extensions connected to our legacy pbx, that is installed on front of asterisk. So : Zapata Calls to SIP extensions 4XXX - OK IAX to SIP 4XXX-OK SIP to SIP 4XXX -