search for: ssrc

hi, i have strange problem with no rtp packets from asterisk after dns query. see pcap below centos6/asterisk 13.9 + chan_sip 172.23.0.3 - asterisk 172.23.5.1/2 - voip phones any ideas/hints? 1170 25.028206000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49318, Time=1442112256 1171 25.045556000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4468, Time=716240 1172 25.045629000 172.23.0.3 -> 172.23.5.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x3566361, Seq=60990, Time=716240 1173 25.048134000 17...

Audio packets are running... 961 16.150421076 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x6A3E0AF1, Seq=28402, Time=73280 962 16.170411284 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x6A3E0AF1, Seq=28403, Time=73440 963 16.190381989 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x6A3E0AF1, Seq=28404, Time=73600 964 16.210387990 192.168.5.15...

...] DEBUG[11426] res_fax.c: channel 'SIP/SOV20001-0007cb04' using FAX session '660' [Oct 14 14:16:17] DEBUG[11426] chan_sip.c: T38 state changed to 3 on channel SIP/SDSD0005-0007cb05 PCAP text output of 1st case: 216.025063 192.168.196.3 -> 192.168.196.94 RTP PT=ITU-T G.711 PCMA, SSRC=0x6118CC28, Seq=4877, Time=1179871936 216.042133 192.168.196.94 -> 192.168.196.3 RTP PT=ITU-T G.711 PCMA, SSRC=0x205734A, Seq=8641, Time=2227760 216.045031 192.168.196.3 -> 192.168.196.94 RTP PT=ITU-T G.711 PCMA, SSRC=0x6118CC28, Seq=4878, Time=1179872096 216.062169 192.168.196.94 -> 192.1...

...Gateways of the carrier, the media gateway however sends RTP with a completely different timestamp to Asterisk B, so Asterisk B copies that timstamp and Asterisk A gets an audio hickup. IE asterisk B recieves: asterisk B sends to A sequence 1 timestamp 0 SSRC 1234 from ip 1.2.3.4 sequence 1 timestamp 0 SSRC 4321 from ip Asterisk B sequence 2 timestamp 30 SSRC 1234 from ip 1.2.3.4 sequence 2 timestamp 30 SSRC 4321 from ip Asterisk B sequence 3 timestamp 60 SSRC 1234 from ip 1.2.3.4 sequence 3 timestamp 60 SSRC 4321...

...December/000693.html saying that the PR 12544 is not really the corresponding issue :) The correct one is PR 13392: http://gcc.gnu.org/bugzilla/show_bug.cgi?id=13392 Interesting fact is that -O2 (or -O3) goes somehow around this problem. It looks like -O2 is not used here: /home/vak/ssrc/llvm/mklib --tag=disable-shared --silent --mode=compile g++ -c -I/home/vak/ssrc/llvm/lib/Support -I/home/vak/ssrc/llvm/lib/Support -I/home/vak/ssrc/llvm/include -I/home/vak/ssrc/llvm/include -I../../include -I/home/vak/ssrc/llvm/include -D_GNU_SOURCE -D__STDC_LIMIT_MACROS -Wall -W -Wwrite-strings...

Hi Everyone, We're routing calls through Asterisk (dialing in via sip and then dialing out via SIP). We've noticed a curious behavior in chan_sip that doesn't persist with chan_pjsip. When examining the packet capture, we're seeing the SSRC changing constantly on the call. At first it happens over a variable interval (15s 6s etc) but eventually it ends up changing exactly every 1000ms. Every time the SSRC changes, it causes a very minor but noticeable gap in audio. The fact that it's changing on this exact interval makes me thi...

Evnin... Googled around for this strange error meesage with no helpful results at all... Does somebody has any idea what this means? "Forcing Marker bit, because SSRC has changed" At the same time I only get inbound audio but other side can't hear me...sometimes I just hear my echo and nothing from other side... Asterisk version 1.2.9 and both participants with public IP addresses...so no NAT/Firewall involved... cheers rick

Dear list (and more specifically Bret), I am getting one-way (inbound only) audio when trying to place a SIP call via voip.trxtel.com (i.e. 18005558355@voip.trxtel.com). The Cli spits out "== Forcing Marker bit, because SSRC has changed" 5 times after atempting a native bridge. I realize this is most certainly a NAT issue, the * server is behind one. Sip.conf has externip=, and localnet=. Can someone explain what SSRC changing implies is going on, and why it affects NAT? This is actually the first NAT issue I hav...

...eneration 0 // a=ice-ufrag:QJy1Fslu8ITGYl/d // a=ice-pwd:Q8N6+0PPj4CUG6leGAie7zaL // a=ice-options:google-ice // a=fingerprint:sha-256 CF:30:A7:7F:71:11:D4:5E:B0:E7:E3:F9:D8:C2:F4:60:2A:D0:76:46:F8:3A:97:01:C9:0C:5A:F7:B8:7D:C1:43 a=setup:actpass a=mid:audio // a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level // a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time // a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap...

...ple, I have a file with 3 streams, but the weird stuff is that 2 streams are the same (e.g., have the same source address and port and same destination address and port). Example: "Source Address","Source Port","Destination Address","Destination Port","SSRC","Payload","Packets","Lost","Max Delta (ms)","Max Jitter","Mean Jitter","Status" "6X.XXX.XXX.XXX",34170,"1XX.XXX.XXX.XXX",10602,277011456,"g711A",7289,0,21. 303999999996449,21.265543809819981,0....

...IP/209.120.202.94:5060-0533 is making progress passing it to SIP/1000-c317 -- SIP/209.120.202.94:5060-0533 answered SIP/1000-c317 -- Attempting native bridge of SIP/1000-c317 and SIP/209.120.202.94:5060-0533 Jun 3 22:56:09 WARNING[28769]: rtp.c:500 ast_rtp_read: Forcing Marker bit, because SSRC has changed However the calls complete correctly. I'm using 1.2.8 asterisk stable release. what does that mean? Thanks, Erick.

...answers, for a few seconds (4/5 sec typical) some SIP client could not hear anything (the ringing was heard well!), then the audio comes back again with no problem. Looking for any differences between the behaviour of version 1.4.17 and 1.4.19 I found that in the new version the RTP stream changes SSRC between the early media session and the actual call session. This seems to me quite pretty, and a major part of SIP clients seems not to be disturbed by it. Anyway I'd like to ask you a couple of things on this issue: 1) Is the changing of ssrc standard compliant? (I suppose yes, because the s...

...should be, but sometimes while the asterisk plays MOH (or somebody transfers call to another extension) the timestamps on RTP packets will fall to past. Providers media gateway dosn't like that. The marker bit is correctly set but it seems like that dosn't change anything. Sequences and SSRC-s are Ok, no packet loss. Has anyone seen something like this before and knows what is the cause and how to fix this? I've tried many changes in config and upgraded to 1.6.1 but it didnt change anything, currently running asterisk 1.4.26.1 on 64 bit intel platform with opensuse. Here is the...

...6 0 typ host generation 0 a=ice-ufrag:l8AWdK4ft+AnAYGl a=ice-pwd:3tLKvT97tf0GQr+e8v8bKncd a=ice-options:google-ice a=fingerprint:sha-256 89:D4:EB:6E:9C:41:36:03:A1:44:CD:A2:08:78:CD:86:FE:EC:30:09:53:0F:77:CE:BA:8E:DE:8C:1B:A1:41:10 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:dYMEPP1zoNS/W70Ln6cnBCtHXDCq6ciLZmHDHdFj a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Gr23SpFGDiukOKyrrfAauWssQ+3pYjD0jwkK9hOo a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000...

Hi, I'm getting messages like res_srtp.c:395 ast_srtp_unprotect: SRTP unprotect failed with replay check failed (index too old), retrying == SRTP unprotect failed on SSRC 576693764 because of authentication failure 10 == SRTP unprotect failed on SSRC 576693764 because of authentication failure 160 [...] ... after a couple minutes during voice calls after which the connection is being aborted. It seems that not all connections are affected but only a few, though...

...le [May 10 10:45:24] a=ice-options:trickle [May 10 10:45:24] a=fingerprint:sha-256 C9:33:B0:E9:7C:F4:F2:39:98:A6:5C:AE:16:7F:5E:18:99:8F:9F:EB:DC:C6:E3:D5:EA:5B:AE:CD:DE:75:79:0B [May 10 10:45:24] a=setup:actpass [May 10 10:45:24] a=mid:audio [May 10 10:45:24] a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level [May 10 10:45:24] a=sendrecv [May 10 10:45:24] a=rtcp-mux [May 10 10:45:24] a=rtpmap:111 opus/48000/2 [May 10 10:45:24] a=rtcp-fb:111 transport-cc [May 10 10:45:24] a=fmtp:111 minptime=10;useinbandfec=1 [May 10 10:45:24] a=rtpmap:103 ISAC/16000 [May 10 10:45:24] a=rtpmap:104 ISAC/32000...

hey all, can any body know what this parameter stands for i got RTPAUDIOQOS while i have SIP channels but could not understand then what this parameter tell * ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.000000;txcount=83;rlp=0;rtt=14818.715000 * if any one know plese help me to or give any documentation link regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium...

If this has been covered before - I appologize. We use some Sipura SPA-2000's with the g711 codec and all seems fine (except for the occasional failure to register errors in my asterisk logs - but I will save that for another post). g711 call quality is on par with our Cisco 7960's. However, when using the g729 codec, the call quality on the Sipura device goes downhill on the PSTN side

....168.122.1 0 typ host generation 0 ????a=ice-ufrag:iZyQ8Egkyi8hPKah ????a=ice-pwd:kQ91vFMHVr2lOkZfjGDLSfO+ ????a=ice-options:google-ice ????a=fingerprint:sha-256 81:DE:7E:6F:2B:88:8F:F3:30:82:92:DF:CB:FC:4B:63:BB:2E:BA:85:48:2B:B5:A6:C3:50:A1:42:E4:69:0E:91 ????a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level ????a=sendrecv ????a=mid:audio ????a=rtcp-mux ????a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:haR/UikskQr/AIrry5udqINI1hYfc5zY2I6jrkKT ????a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:waQfKIHI9UyjPVI0vrcUREDbSVZdtfCtRQK71/Ks ????a=rtpmap:111 opus/48000/2 ????a=fmtp:111 minptime=10 ????a=rtpm...

Building PowerPC.td register information header with tblgen Included from PowerPC.td:22: Parsing PowerPCInstrInfo.td:53: Variable not defined: 'GPRC'! make[3]: *** [PowerPCGenRegisterInfo.h.inc] Error 1 make[3]: Leaving directory `/pool/tmp/ssrc/llvm/lib/Target/PowerPC' maybe I just have to "make clean" and/or ./configure BTW, would it be nice to put Depend, Release and Debug into .cvsignore for other llvm fans too? --- Valery A.Khamenya