Displaying 20 results from an estimated 5000 matches similar to: "How to make * send RTCP reports"
2003 Nov 18
1
Will Asterisk be supporting RTCP XR in the future?
This article below came up on the newwire. The RTCP XR RFC was published.
Will Asterisk be supporting this function in a future release? Does anyone
know if any phone vendors are going to be supporting it?
Thanks
Lee Goodman
Our Technology Update this week is about one of those
mechanisms. Known as RTP Control Protocol Reporting Extensions
(RTCP XR), the technology defines a standard way to
2008 Nov 28
1
RTCP too short
Dear Sir,
I'm running Asterisk 1.4.21.2 on a CentOS machine....When running asterisk
-rvvvvv I can see a lot of messages about RTCP too short...
-- Remote UNIX connection disconnected
[Nov 28 13:33:00] WARNING[24863]: rtp.c:891 ast_rtcp_read: RTCP Read too
short
[Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too
short
[Nov 28 13:33:00] WARNING[19803]: rtp.c:891
2010 Jan 28
2
rtp.c:883 ast_rtcp_read: RTCP Read too short
Hi:
I have a linksys voip gateway connecting to an asterisk server ,when i dial a call from the linksys gateway to asterisk , i see repeated messages of a RTP errors ,and at same time i hear fake ring on the linksys?, This is wht i see on asterisk console?:
?
-- Executing [9613070741 at direct:1] Set("SIP/03070741-088bd470", "CALLERID(number)=96170707070") in new stack
??? --
2010 Apr 02
1
RTCP How to stop
Dear all;
I want to stop RTCP from Asterisk-server to phone.
But I want to use RTP.
I looked rtp.conf/sip.conf, but I can't know about it.
Please tell me how to stop RTCP only.
Because , when I access under NAT, my gateway shutdown the port as gateway received RTCP from server.
I use Asterisk 1.6.2.6 or 1.4.29 .
Also SIP/RTP.
thx.
2009 Jan 12
1
RTCP SR transmission error, rtcp halted
Hi,
While looking for the cause of disturbance in call I found this error
coming in console
RTCP SR transmission error, rtcp halted
Google search only shows some bug reports relating to MOH and Hold.
What could cause this message? Could this be a symptom causing call
disturbance? Where should I start digging to find out the reason for
this error?
I am using Asterisk 1.4.19 with zaptel 1.4.9.2
2011 Jan 23
1
RTCP packets when on hold
Hi,
It seems that asterisk doesn't send RTCP packets when a call is on hold. Is there any way to get asterisk to send these packets?
I'm in the process of setting up a Lync (microsoft voice) server which will use an asterisk box as a gateway. The trunking between asterisk and lync is 'working' however when a call is put on hold asterisk stops sending RTCP packets to lync, and
2012 Feb 16
2
Asterisk && RTCP
Hello list,
I need to know about Asterisk's friendly nature with RTCP. I've phones
which support RTCP and they connect to the outer world via multiple
carriers. In one of my recent packet traces I've observed that the caller
initiated a call with rtcp string in SDP while for the same
call dialling our from Asterisk to the carrier has no RTCP string in SDP !
Can anyone please tell why
2001 Feb 14
2
RTP/RTCP payload?
(hello all, this is my first writing. so please
bear with me if I'm wrong anywhere.)
orry to break too lately, but how is the RTP payload
submission is going?
could we see the new payload at March IETF?
I agree that it would be fairy straightforward to
make an RTP payload for ogg vorbis, assuming raw
packets, AFAIK. using physical bitstream is, in
this case, not adequate by the reasons in
2011 Oct 14
3
[Bug 757] New: SIP connection helper not setting RTCP conntrack expectation
http://bugzilla.netfilter.org/show_bug.cgi?id=757
Summary: SIP connection helper not setting RTCP conntrack
expectation
Product: netfilter/iptables
Version: linux-2.6.x
Platform: i386
OS/Version: Ubuntu
Status: NEW
Severity: normal
Priority: P5
Component: ip_conntrack
2019 May 10
4
Asterisk 13.26.0 webRTC: Asterisk not passing along video
Hello
I am trying to set up webRTC video calls from my Chrome webbrowser
(Fedora) to my Chrome webbrowser (Windows 10).
There is local video input (I can see myself), but never video on the
receiving side.
This is the case in both directions (so it makes no difference which
peer is calling which peer).
Both webRTC SIP peers have opus and H264 codec in their peer definition :
Video
2014 May 12
1
SIP call control via RTCP
Hello,
We are using Asterisk 1.4 as call distribution system with simple queues
for SIP calls.
With high load (4000 calls/hour) some calls remain in queue forever (until
queue's max wait time) in spite of being hung up already by the caller. It
seems that when a BYE is lost, Asterisk has no mechanism to check whether a
call is still active.
Is there a way to activate a RTCP call control,
2008 Apr 08
3
RTCP not being sent when on hold
Hello,
When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 and I
place the call on hold, the call is dropped after 30 seconds.
It looks like there is no RTCP/RTP sent to the client from Asterisk while on
hold (music on hold playing to caller) thus client disconnects the call.
During this time, I get the following messages in the CLI:
NOTICE[24194] rtp.c: Unknown RTP codec 126
2009 Oct 01
1
RTP Delayed during RTCP
Hello,
Has anyone encountered that when Asterisk sends RTCP messages, it stops
sending RTP packets until it gets an answer?
Can that be fixed?
Thanks.
2008 Feb 07
1
SIP / RTCP statistics logging
G'day. I am wanting to find out how my SIP service is performing with
Asterisk, especially jitter and dropped packets.
I can get an overview of that using the 'rtcp stats' function at the
console, but is there any way to get those logged to a file or some
other permanent record?
Nothing in logger.conf seems applicable, save perhaps directing verbose
messages somewhere, but it
2008 Nov 07
1
is it possible to deactivate RTCP?
Hi!
Is it possible to deactivate RTCP? (I am using 1.6)
thanks
klaus
2017 Nov 14
2
RTCP + Stasis causing high memory consumption
Hello Asterisk list,
I've facing a memory allocation issue that happens occasionally but on a
consistent basis.
The problem happens as follow, suddenly Asterisk starts consuming a lot of
memory, in a rate of more than 1GB per hour. Kernel will eventually kill it
via the OOM killer when memory is really exausted... This situation does
not generate backtrace because Asterisk is responsive
2017 Dec 13
0
AST-2017-012: Remote Crash Vulnerability in RTCP Stack
Asterisk Project Security Advisory - AST-2017-012
Product Asterisk
Summary Remote Crash Vulnerability in RTCP Stack
Nature of Advisory Denial of Service
Susceptibility Remote Unauthenticated Sessions
Severity
2013 Dec 02
1
Not able to get remote channel variables containing RTCP values
I am not sure if its just me, but i am able to get only local channel
variables containing RTCP QOS values.
The Version is 1.8.14.
I want to store values of bridged channel in CDR.
Phone is Cisco 7941 SIP and with sip show channelstats i see all the
relevant information (jitter,packet loss) i want to get. It even
calculates packet loss in %. But i am not able to store it to CDR.
Asterisk 1.4
2007 Oct 11
0
Understanding RTCP in Asterisk
X-ECN Telecoms-MailScanner-Information: Please contact ECN Telecoms for more information
X-ECN Telecoms-MailScanner: Found to be clean
X-ECN Telecoms-MailScanner-From: yusuf at ecntelecoms.com
X-Spam-Status: No
My third try, humph!
Yusuf wrote:
> Hi,
>
> I am trying to understand the RTCP stats in Asterisk.
>
> 1. I am using the 'h' exten to store the RTCP records in
2017 Sep 19
0
AST-2017-008: RTP/RTCP information leak
Asterisk Project Security Advisory - AST-2017-008
Product Asterisk
Summary RTP/RTCP information leak
Nature of Advisory Unauthorized data disclosure
Susceptibility Remote Unauthenticated Sessions
Severity Critical