similar to: RTP packets send, but no audio

Displaying 20 results from an estimated 20000 matches similar to: "RTP packets send, but no audio"

2019 Dec 12
2
asterisk pjsip webrtc rtp to private IP
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
2019 Dec 12
2
asterisk pjsip webrtc rtp to private IP
with wireshark i need decrypt traffic every call which is time consuming. get debug from pjnat through asterisk is not possible because of technical reasons or nobody did it? in my case its strange that ice candidates are the same good call v=0 o=- 3669976329745317845 2 IN IP4 127.0.0.1 s=- t=0 0 a=msid-semantic: WMS EoNIdKcMZvWBLULGqGPJTDe12ujjFEemeapo m=audio 52421 RTP/SAVPF 8 0 101 c=IN
2020 Sep 22
2
Negotiates g729 but RTP contains g711
Hi, We have a scenario where inbound calls from an upstream provider (chan_sip) sent downstream (chan_iax2) negotiates only g729 yet RTP media contains g711. Both the upstream and downstream trunks are limited to only offering g729 whilst the initial invite from our upstream provider offers both g711 and g729. Asterisk presumably simply forwards the media from iax2 trunk encapsulation to sip
2020 Sep 24
2
Negotiates g729 but RTP contains g711
Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing
2015 Nov 12
3
No sound with internal calls depending on which phones
Dear all, I have a very strange problem : * external calls work perfectly, * internal calls between some phones too, * but internal call between two similar phones don't work !!! (Snom 710) When we have sound, there are no errors in asterisk. When we do not have sound, there is the following error : * [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP module
2015 Nov 12
3
No sound with internal calls depending on which phones
Snom default configuration is SRTP enabled. You should disable the SRTP from the phone web GUI configuration Sincerely, Sam Basan From: Mitul Limbani [mailto:mitul at enterux.in] Sent: Thursday, November 12, 2015 5:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] No sound with internal
2016 Aug 10
2
Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
Hello thank you for your answer. I don't understand how there are many tutorials and examples on the web where every time the outcome is a working setup. Very strange I feel now after my personal experience with Asterisk 11 and webRTC. You also say Asterisk 13. How about Asterisk 12 then ?? Kind regards. On 10-08-16 21:53, Matt Fredrickson wrote: > I don't see an ice-ufrag or
2020 May 17
1
PJSIP sending RTP to private address
My phone is located behind a NAT, 172.16.0.0/21. Asterisk 16 is on a public IP. PJSIP has the config below: force_rport=yes direct_media=yes disable_direct_media_on_nat = yes direct_media_method=invite But when I send a call I see the RTP being sent to my private address, vs the public IP. This only happens when Asterisk has dialed the call to another carrier. If instead of Dial I choose
2009 Jan 20
2
SIP DTMF problem with SNOM
Hi! I have two identical SIP accounts on Asterisk 1.4.22. One account is registered with eyebeam, the other one is registered with a SNOM phone. When using the eyebeam client DMTF detection works fine, when using the SNOM phone many digits are missing in the DTMF detection. I analyzed with wireshark and both phones uses RFC 2833 and the trace looks pretty the same. Also the rtp debug log
2009 Jan 20
5
the FXS ports of Digium and damaging if connected to Tel Line
Hi All; I am facing a problem that always the users confused and connect the telephone line coming from the telephone service provider to the FXS port and cause it to be damaged, specially if the card was 2 fxs and 2 fxo, so they make mistake and connect the line to fxs while it should be connected to fxo. What is the solution for this disaster? Regards Bilal
2023 Apr 17
1
RTP address learning and timing problem
It's probably best if you read the logic[1]. There's an entire comment that talks about how it works. [1] https://github.com/asterisk/asterisk/blob/20/res/res_rtp_asterisk.c#L8158 On Mon, Apr 17, 2023 at 7:10 PM David Cunningham <dcunningham at voisonics.com> wrote: > Hi Joshua, > > Could you confirm if the 5 second period for learning a new audio stream > is a minimum
2012 Dec 20
2
asterisk 11 and no RTP
I have a CentOS 6.3 machine I installed Asterisk 11, worked fine... I then tried to install on Cents 5.8, seemed to go fine... Then when I placed a call I got this: ast_rtp_instance_new: No RTP engine was found. Do you have one loaded? Did a search and found issues with ARM and this problem but did not help me, not using gtalk or anything. Just call between two polycom phones on local network.
2023 Apr 17
1
RTP address learning and timing problem
Hi Joshua, Thank you for that. From the code it kind of looks like STRICT_RTP_LEARN_TIMEOUT is a minimum, not a maximum: if (!ast_sockaddr_isnull(&rtp->strict_rtp_address) && STRICT_RTP_LEARN_TIMEOUT < ast_tvdiff_ms(ast_tvnow(), rtp->rtp_source_learn.start)) { ast_verb(4, "%p -- Strict RTP learning complete - Locking on source address %s\n", Our call shows: #
2023 Apr 18
1
RTP address learning and timing problem
I don't know in that specific output what happened. Your best course of action is to add further logging or step through the logic with all of the knowledge you have of the RTP streams to understand what is happening. On Mon, Apr 17, 2023 at 8:52 PM David Cunningham <dcunningham at voisonics.com> wrote: > Hi Joshua, > > Thank you for that. From the code it kind of looks like
2019 Dec 12
2
asterisk pjsip webrtc rtp to private IP
Asterisk is on public IP (as described in the first email) i have 10 years experience in voip, 4 years webrtc in production. i know about ICE/STUN/DTLS-SRTP. yes, not every detail but the basic mechanism but i confess. i dont understand WHY Asterisk SOMETIMES switches destination IP in RTP. this is not only about ICE. its about RTP engine too which is Asterisk specific and Asterisk DEBUG is
2008 Mar 20
1
Newbie: Two problems with Asterisk Config, Please Help
Hi, I am sorry my questinos are too fundamental. I am new to Asterisk, and hope to catch up as fast as I can. Problem 1: I have my SIP client ( in one PC .102) and SIP server ( in another PC .101) within the same land. They can make SIP connection, but when the SIP client makes call to play an audio file, I can only hear a "beat" sounds, and then nothing else. In the console, I can
2014 Oct 23
1
Auto video call hangup
Hi, I use a simple scheme: SIP video phone A (h264/Asterisk 1.8.11) <---IAX2 trunk----> SIP video phone B (h264/Asterisk 11.7.0) When calls from A to B and vice versa drop on pickup. On B side: [Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the marker bit
2023 Apr 14
2
couldn't allocate a port for RTP instance
Hello, my logs are flooded with: WARNING: The 'stasis/m:cdr:aggregator-00000005' task processor queue reached 5000 scheduled tasks again. and then, when call came, I got this: ERROR: Oh dear... we couldn't allocate a port for RTP instance '0x6e1e680fd670' WARNING: Unable to cancel schedule ID 0. This is probably a bug (res_rtp_asterisk.c: dtls_srtp_stop_timeout_timer,
2011 May 12
1
Different IP addresss for SIP and RTP
Hello, is it possible to set an IP address for RTP different than the one used for SIP? I want to use asterisk behind a sip proxy (opensips), but I was thinking if I could avoid having to run rtpproxy on the sip proxy server and let asterisk itself take care of it. So that: Asterisk SIP address : local ip address Asterisk RTP address : global ip address regards, takeshi -------------- next
2007 Dec 03
1
Subject: Newb Question
Hi, Use orecx, voip call recording and monitoring. www.orecx.com Thanks & Regards, Vidura Senadeera, Sri Lanka. Tel - +94114520001 Mobile - +94777766596 yahoo/skype Ids - vidurased > ------------------------------ > > Message: 17 > Date: Fri, 30 Nov 2007 08:58:41 +0530 > From: ram <talk2ram at gmail.com> > Subject: Re: [asterisk-users] Newb Question > To: