search for: 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 ! C...

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 ast_rtcp_read: RTCP Read too short [Nov...

...ion :   Video Support: Yes   Prim.Transp. : WS   Allowed.Trsp : WSS   SIP Options  : (none)   Codecs       : (opus|h264)   Status       : OK (75 ms)   Useragent    : SIP.js/0.12.0   Reg. Contact : sip:llghjqha at 192.0.2.239;transport=wss   RTP Engine   : asterisk   Encryption   : Yes   RTCP Mux     : Yes   Video Support: Yes   Prim.Transp. : WS   Allowed.Trsp : WSS   SIP Options  : (none)   Codecs       : (opus|h264)   Status       : OK (47 ms)   Useragent    : SIP.js/0.12.0   Reg. Contact : sip:6ltm4mqe at 192.0.2.7;transport=wss   RTP Engine   : asterisk   Encryption   :...

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...

...] Set("SIP/03070741-088bd470", "CALLERID(number)=96170707070") in new stack ??? -- Executing [9613070741 at direct:2] Dial("SIP/03070741-088bd470", "SIP/usa/9613070741") in new stack ??? -- Called usa/9613070741 [Jan 28 18:17:36] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short [Jan 28 18:17:42] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short ??? -- Call on SIP/usa-08906450 left from hold ??? -- SIP/usa-08906450 is making progress passing it to SIP/03070741-088bd470 ??? -- SIP/usa-08906450 is ringing ??? -- Call on SIP/usa-08906450 l...

...ying with * for 10 days now. I am testing with a couple of vocaltec h.323 gateways (FXO and PRI) cisco ata-186 (configured for SIP) and MSN messenger (SIP). They all seem to interoperate. However I have a problem when * is sending calls to the vocaltec gateways. Vocaltec gateways are monitoring the RTCP reports send from the remote gateway (in this case *) and if they don't get a report for 60 seconds they will disconnect the call (assuming internet disconnection). Because of this all my calls have duration of one minute. I can see on the console that * is detecting incoming RTCP reports s...

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 (RTC...

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 Moderate Exploits Known No...

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 received from '0.0.0.0' In sip.conf I have rtpkeepalive=15 but that...

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 S...

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 a...

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 AssignedTo: netfilt...

...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 RFC-1889. but I don't think that's enough. rather than sending comments in the same RTP packet, we'd better send it in RTCP packet. to do that we should define an RTCP APP name field for needed situations, or an RTCP extension. (or, could we piggyback on RFC-2793 and rather define an XML format?...) of course, when the tarkin goes beta, we would need to define its own payload, AND the payload for multiplexed physical...

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 CDRS. > However, only if the > caller hangups does the RTCP values have anything in them. If the > caller hangups, the > values gets stored in CDRs, but they all empty(0). So even on the...

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 Exploits K...

[Apr 15 11:12:19] ERROR[2624]: rtp.c:2447 ast_rtcp_write_sr: RTCP SR transmission error to aaa.bbb.ccc.ddd:37259, rtcp halted Operation not permitted [Apr 15 11:12:23] ERROR[2624]: rtp.c:2447 ast_rtcp_write_sr: RTCP SR transmission error to aaa.bbb.ccc.ddd:38563, rtcp halted Operation not permitted What is the specific nature of this traffic?...

...t;IAX2/BOX-YOCAN-10022", "SIP/YOCAN-3STARSNET/0473775006") in new stack [Oct 3 17:40:55] -- Called YOCAN-3STARSNET/0473775006 [Oct 3 17:41:01] -- SIP/YOCAN-3STARSNET-076fa990 is making progress passing it to IAX2/BOX-YOCAN-10022 [Oct 3 17:41:06] ERROR[1499]: rtp.c:2482 ast_rtcp_write_sr: RTCP SR transmission error to 8X.1X.XX.XX:14129, rtcp halted Operation not permitted [Oct 3 17:41:15] == Spawn extension (from-BOX-YOCAN, 0473775006, 3) exited non-zero on 'IAX2/BOX-YOCAN-10022' [Oct 3 17:41:15] -- Hungup 'IAX2/BOX-YOCAN-10022' QUESTIONS : 1. wh...

...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, e.g. Asterisk should hang up when he stops receiving RTCP messages? Kind regards -- *Jan **Gaida* Ingeniero Desarrollo Software C/ Marconi 3 (PTM) 28760 Tres Cantos Spain jan.gaida at grupoamper.com | www.grupoamper.com -- This message and any attachments are intended...

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.

Hello! I want to get rtcp stats to cdr. (btw, I run asterisk 1.6.2.11) There is howto here http://www.voip-info.org/wiki/view/Asterisk+RTCP But I (and my users) do bridged calls from dahdi to sip, so in h extension channel is dahdi , and it doesn't contain rtcp stats. There is info about function shared. But I can'...