search for: fmtp

...ll ; First disallow all codecs allow=siren14 ; Is this the right name? And the INVITE comes from the Polycom softphone with an SDP of: ... User-Agent: Polycom VV 8.0.4.4035. ... m=audio 12386 RTP/AVP 99 98 97 102 101 103 9 15 18 0 8. a=rtpmap:99 SIREN14/16000. a=fmtp:99 bitrate=48000. a=rtpmap:98 SIREN14/16000. a=fmtp:98 bitrate=32000. a=rtpmap:97 SIREN14/16000. a=fmtp:97 bitrate=24000. a=rtpmap:102 G7221/16000. a=fmtp:102 bitrate=32000. a=rtpmap:101 G7221/16000. a=fmtp:101 bitrate=24000. a=rtpmap:103 G7221/16000. a=fmtp:103 bitrate=16000. a=rtpmap:9 G722/8000....

...t?use H.264 codec with following sdp information: Android Based Client SDP Parameters v=0 o=- 1325786904 1325786904 IN IP4 172.16.130.47 s=Polycom RealPresence c=IN IP4 172.16.130.47 b=AS:1920 t=0 0 a=sendrecv m=audio 3230 RTP/AVP 118 115 114 113 0 8 119 a=rtpmap:118 SIRENLPR/48000 a=fmtp:118 bitrate=64000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:114 G7221/32000 a=fmtp:114 bitrate=32000 a=rtpmap:113 G7221/32000 a=fmtp:113 bitrate=24000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:119 telephone-event/8000 a=fmtp:119 0-15 m=video 3232 RTP/AVP 109 110...

...timal, but I really need it, even if I have to patch Asterisk Thanks for your help SDP send by Tandberg : -------------------------- v=0 o=tandberg 1 5 IN IP4 192.168.50.10 s=- c=IN IP4 192.168.50.10 b=CT:1920 t=0 0 m=audio 48260 RTP/AVP 100 101 9 8 0 102 b=TIAS:64000 a=rtpmap:100 G7221/16000 a=fmtp:100 bitrate=32000 a=rtpmap:101 G7221/16000 a=fmtp:101 bitrate=24000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:102 telephone-event/8000 a=fmtp:102 0-15 a=sendrecv m=video 48262 RTP/AVP 97 98 99 34 31 c=IN IP4 192.168.50.10 b=TIAS:1920000 a=rtpmap:97 H264-RCDO/90000 a=fm...

Dan Cropp wrote: > I had my screenshots flipped. Is there a way to make sure the Contact field is NOT included in the ACK response to the OK (for the Answer)? > > PJSIP is including the Contact for the ACK response to the OK. > Contact:<sip:1234 at xxx.xxx.xx.xxx:5060> > There is no configuration option to configure this behavior. What is the full SIP signaling? -- Joshua

Asterisk Project Security Advisory - AST-2018-003 Product Asterisk Summary Crash with an invalid SDP fmtp attribute Nature of Advisory Remote crash Susceptibility Remote Authenticated Sessions Severity Minor Exploits Known No...

...ivacy=off;screen=no Max-Forwards: 70 User-Agent: Asterisk PBX 13.0.0 Content-Type: application/sdp Content-Length: 239 v=0 o=- 540555224 540555224 IN IP4 XXX.XXX.XXX.XXX s=Asterisk c=IN IP4 XXX.XXX.XXX.XXX t=0 0 m=audio 10030 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv <--- Received SIP response (378 bytes) from UDP:64.2.142.93:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport=5060;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a From: "Dan" <sip:291 at XXX.XXX.XXX....

...: Asterisk 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....

...low: INVITE, 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;rp...

...with tcpdump, 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 RT...

Hey i have an interesting topic to discuss here. The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support . the problems that i faced with this is the following and i hope i could get an advise here. asterisk 13 vanilla version has some issues marking the video packets this complain

Hello All, I'am doing a video call between two Video Phones, and i see that Asterisk is stripping the fmtp parameters for the h263 video line in SDP. For example a line similar to the below is stripped, a=fmtp:xx CIF=4;QCIF=2;F=1;K=1 Asterisk is configured NOT to be present in the Media path (My version : Asterisk 1.4.19.1 ). I have the following enabled in my sip.conf. canreinvite=y...

While trying to use direct_media I'm seeing RTP payload mismatches after succesful reinvites. Initial INVITE from endpoint A to asterisk has rfc4733 DMTF m=audio 35648 RTP/AVP 9 8 111 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 >From asterisk to upstream U: m=audio 14338 RTP/AVP 9 8 111 18 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 So the payload types in the RTP streams from A and to U differ. This works fine when asterisk is relaying media. With direct_media=yes there are reinvites sent from a...

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

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.

Hi Folks, thanks for the great work, not sure if this is the right list for this type of quesiton. We are looking to use only Opus as "one codec for all", with VoIP-out obviously we want to tune it. I am planning to use fmtp in SDP to control server/client Opus settings. Something like : - *maxplaybackrate*: a hint about the maximum output sampling rate that the receiver is capable of rendering in Hz. maxplaybackrate=8000 - *maxaveragebitrate*: specifies the maximum average receive bitrate of a sessio...

...EL, OPTIONS, 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=z9hG4bK541d55...

...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 [May 10 10:45:24] a=rtpmap:9 G722/8000 [May 10 10:45:24] a=rtpmap:0 PCMU/8000 [May 10 10:45:24] a=rtpmap:8 PCMA/8000 [May 10 10:45:24] a=rtpmap:106 CN/32000 [May 10 10:45:24] a=rtpmap:...

...Y, INFO, PUBLISH, MESSAGE Content-Type: application/sdp Content-Length: 394 v=0 o=root 15363811 15363812 IN IP4 192.168.2.1 s=sipgate VoIP GW c=IN IP4 192.168.2.1 t=0 0 m=audio 7070 RTP/AVP 8 0 3 97 18 112 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:112 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <--- Transmitting SIP response (733 bytes) to UDP:217.10.79.9:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.10.79.9;rport=5060;received=...

...and G729 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...

...SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 1799 v=0 o=root 469858785 469858785 IN IP4 10.1.0.4 s=Asterisk PBX 11.21.0 c=IN IP4 10.1.0.4 b=CT:384 t=0 0 m=audio 14814 UDP/TLS/RTP/SAVPF 4 0 3 8 18 110 9 101 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=ice-ufrag:0e746cd50c88ce6e383ff3882acebb80 a=ice-pwd:1a9a09862254ae253f06a0...