search for: annexb

Hello Looking the SIP debug we see a change in the SETUP message from the Asterisk 1.0.x version to the 1.2.4. In the 1.2.4 we notice this line: a=fmtp:18 annexb=no This line cause problems in our plattform (We think our SIP -> h323 gateway can't parse this line) Why this line its present in 1.2.4 version? Have anybody some clue? Regards JS.

...sk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:9990@172.30.42.5> Content-Type: application/sdp ontent-Length: 235 v=0 o=root 5641 5641 IN IP4 172.30.42.5 s=session c=IN IP4 172.30.42.5 t=0 0 m=audio 29816 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Executing VoiceMailMain("SIP/eden-1000a-4150cc98", "1000@eden") in new stack -- Playing 'vm-password' (language 'en') pbx*CLI> <-- SIP read from 10.253.4.50:5...

im having a strange problem related to call-limit for peers. well im not sure if its related to call-limmit or not. Bottom line is: I call a user A, from user B. user B hears silence, untill it goes to voicemail. when user B hangsup. user B's call limit is reset to 0 but user A's call limit is not reset.strange thing is user A's status on cli is shown as NOANSWER, while user B did not

...dump, I have seen that all the successful calls have SDP negotiation that look like this: (inside INVITE request body from SIP carrier) v=0 o=msw.chance4minutes.net 1234 0 IN IP4 38.126.208.46 s=sip call c=IN IP4 38.126.208.46 t=0 0 m=audio 30552 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 (inside 200 OK response body from asterisk) v=0 o=root 835643920 835643920 IN IP4 201.234.196.171 s=Asterisk PBX 11.10.0 c=IN IP4 201.234.196.171 t=0 0 m=audio 12112 RTP/AVP 0 8...

...ITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 291 v=0 o=root 21520 21520 IN IP4 151.196.61.115 s=session c=IN IP4 <public IP> t=0 0 m=audio 11968 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv -- Called bw_outbound/+18885551212 FreePBX*CLI> <--- SIP read from 216.82.224.202:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP public IP:5060;branch=z9hG4bK6ea30a1a;rport=5060 F...

...4. Content-Length: 449. . v=0. o=- 6 2 IN IP4 10.0.0.25. s=CounterPath eyeBeam 1.5. c=IN IP4 10.0.0.25. t=0 0. m=audio 4148 RTP/AVP 98 18 3 101. a=alt:1 3 : 6ceGNvpQ aNAT7Mk6 10.0.0.25 4148. a=alt:2 2 : cu+cL3mB rdqEXGtX 192.168.132.1 4148. a=alt:3 1 : uoim9Hbs Eiu4Y4zw 192.168.80.1 4148. a=fmtp:18 annexb=yes. a=fmtp:101 0-15. a=rtpmap:98 iLBC/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. a=x-rtp-session-id:E06A42E19E7244AFBF10DCAF883B488B. # U 10.201.0.224:5060 -> 10.0.0.25:2750 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 10.0.0.25:2750;branch=z9hG4...

...ONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 279. . v=0. o=root 1631923320 1631923320 IN IP4 192.168.2.10. s=EXAMPLE Systems. c=IN IP4 192.168.2.10. t=0 0. m=audio 52034 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 2014/08/12 07:34:20.903830 192.168.2.10:5060 -> 192.168.2.20:5080 INVITE sip:873359633037 at 192.168.2.20:5080 SIP/2.0. Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK541d5594;rport....

...9 in the INVITE and they say that I have to ignore T38 and use G729 in the voice call. When a fax call is made Movistar send only T38 in the INVITE. Invite example: v=0 o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2 s=sip call c=IN IP4 192.168.1.2 t=0 0 m=audio 6370 RTP/AVP 18 101 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 m=image 6372 udptl t38 a=T38FaxVersion:0 a=T38FaxMaxBuffer:1100 a=T38FaxMaxDatagram:612 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy How can I ignore T38 and use only G729 for this call?...

...STER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: multipart/mixed ;boundary=unique-boundary-1 Content-Length: 770 --unique-boundary-1 Content-Type: application/SDP v=0 o=- 4162 1 IN IP4 172.25.103.222 s=- t=0 0 m=audio 5234 RTP/AVP 18 8 0 c=IN IP4 172.25.103.229 a=fmtp:18 annexb=no a=ptime:30 a=sendrecv --unique-boundary-1 Content-Type: application/x-nt-mcdn-frag-hex ;version=sse-4.50.88 ;base=x2611 Content-Disposition: signal ;handling=optional 0500b201 0107130081900000a2 09090f00e9a08300010032 131e070011fd1800a1160201010201a1300e8102010582010184020000850104 1315070011f...

...a): rtpmap:3 GSM/8000 Media Attribute Fieldname: rtpmap Media Attribute Value: 3 GSM/8000 Media Attribute (a): rtpmap:18 G729/8000 Media Attribute Fieldname: rtpmap Media Attribute Value: 18 G729/8000 Media Attribute (a): fmtp:18 annexb=no Media Attribute Fieldname: fmtp Media Attribute Value: 18 annexb=no Media Attribute (a): rtpmap:111 G726-32/8000 Media Attribute Fieldname: rtpmap Media Attribute Value: 111 G726-32/8000 Media Attribute (a): rtpmap:8 PCMA/8000...

...=calling;counter=0;npi=NPI_E164;ton=TON_NATIONAL P-Asserted-Identity: <sip:3x8x6x3x3x at 10.39.1.19;user=phone> Allow: UPDATE,REFER Content-Length: 293 v=0 o=- 0 138163748 IN IP4 xx.yy.xx.yy s=IMSS e=unknown at invalid.net c=IN IP4 xx.yy.xx.yy t=0 0 m=audio 43718 RTP/AVP 8 18 3 101 a=fmtp:18 annexb=no a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sqn: 0 a=cdsc: 1 image udptl t38 <-------------> <--- SIP read from UDP:xx.yy.xx.yy:5060 ---> INVITE sip:5x5x7x9x0x3 at 192.168.1.168:5060 SIP/2.0 Via: SIP/2.0/UDP xx.yy.xx.yy:5060;branch...

...NFO Supported: replaces, timer Contact: <sip:6615xxxxx at 130.117.xxx.xxx> Content-Type: application/sdp Content-Length: 235 v=0 o=root 1698171141 1698171142 IN IP4 130.117.xxx.xxx s=Asterisk PBX 1.6.1.18 c=IN IP4 130.117.xxx.xxx t=0 0 m=audio 39124 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Packet2Packet bridging SIP/800902-00001794 and SIP/130.117.110.21-00001795 >>> ATA ACK's the OK message: <--- SIP read from UDP://82.158.83.xxx:5062 ---> ACK sip:6615xxxxx at 130.117.xxx.xxx SIP/2...

...: <sip:204 at 41.146.208.131> Content-Type: application/sdp Content-Length: 337 v=0 o=root 777980638 777980638 IN IP4 41.146.208.131 s=Asterisk PBX 1.6.2.13 c=IN IP4 41.146.208.131 t=0 0 m=audio 19726 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- ? Retransmitting #5 (no NAT) to 10.168.7.103:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.168.7.103:5060;branch=z9hG4bK-60ef482f;received=10.168.7.103 From: Ridge <sip:287 at 10.10.0.1>;tag=f3...

...ion/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 522 v=0 o=testacc77005 8004 8000 IN IP4 192.168.1.104 s=SIP Call c=IN IP4 192.168.1.104 t=0 0 m=audio 5020 RTP/SAVP 0 8 18 9 2 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:8m7ZfG+0t3KBFGK40IfDO11SZ6D54glKKIwdgo00|2^32 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:nn+id/sSK7OErMfnZZduKNPLejpscxx1vUQB2seO|2^32 <--- Reliably Tran...

...,norefersub Content-Length: 274 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 1914 0 IN IP4 192.168.1.102 s=SIP Call t=0 0 m=audio 25584 RTP/AVP 0 8 18 101 c=IN IP4 192.168.1.102 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/0 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- (16 headers 13 lines)--- Using INVITE request as basis request - 00115cd9-d0370002-799a069f-51955597@192.168.1.102 Sending to 192.168.1.102 : 5060 (non-NAT) Reliably Transmitting (NAT) to 68.5.xxx.xxx:51065: SIP/2.0 407 Proxy Auth...

...ntent-Type: application/sdp. > Content-Length: 312. > . > v=0. > o=root 1860889533 1860889534 IN IP4 192.168.2.10. > s=Asterisk PBX UNKNOWN__and_probably_unsupported. > c=IN IP4 192.168.2.10. > t=0 0. > m=audio 60646 RTP/AVP 18 101. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > ACC: transaction answered: > timestamp=1365524222;method=INVITE;from_tag=FCA0BFC0-B585477D;to_tag=as0a76fcde;call_id= > 595ad334-f06e97fa-3bbc8137 at...

Hi All, I have a question regarding the nominal jitter buffer configuration: The call was setup as G.729A (annexb=no), ptime=20ms and nominal jitter buffer size = 50ms, and round trip delay is 200ms, the TDM side will experience intermittent one way voice during the call, but IP side can always heard the voice from TDM side. My question is, should this possible caused by the nominal jitter buffer? If there did...

...ssues? In addition I can say that we are using a quite large jitter buffer in zapata.conf: jitterbuffers=16 (=> 0.32s) Moreover, it uses the fixed implementation, because when I tried the adaptive one I experienced one-way audio. Finally I have to note that, using a Siemens IP phone (G.729 no AnnexB) in conditions of no load on servers, I could replicate non-deterministically (sigh!) each of these problems, with a very noisy audio, and a annoying period of silence during the first seconds of call. Regards, Francesco PS. Previous versions of asterisk and zaptel presented an identical situatio...

...#39;t Asterisk support UNENCRYPTED_SRTCP as crypto parameters in sdp? FYI SDP looks like this. v=0 o=- 1429194215 1 IN IP4 XX.XX.XX.XX s=- c=IN IP4 XX.XX.XX.XX b=TIAS:64000 t=0 0 a=avf:avc=n prio=n a=csup:avf-v0 m=audio 50096 RTP/SAVP 0 18 120 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:120 telephone-event/8000 a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:zUVSWsFB/WjVtLxXojBT7zbNvuQ4BkOwcCkD/AjM|2^20 UNENCRYPTED_SRTCP And on CLI I see, DEBUG[1568][C-00000000] sip/sdp_crypto.c: local_key64 7vXot5kn/sl/GYv5ENN6yW0PZZapQ00c++biLgoX len 40 WARNING[1568][C-0000000...

...rds: 70 Date: Sat, 24 Jun 2006 16:12:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 235 v=0 o=root 3131 3131 IN IP4 192.168.0.1 s=session c=IN IP4 192.168.0.1 t=0 0 m=audio 12580 RTP/AVP 18 101 a=rtpmap:18 H723/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- asterisk1*CLI> Retransmitting #1 (no NAT) to 192.168.0.254:5060: INVITE sip:165622270602000@192.168.0.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK45e00b01;rport From: "1656222" <sip:1...