search for: rtpmap

...te: Mon, 16 Jan 2006 11:21:10 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Content-Type: application/sdp..Content-Length: 463....v=0..o=root 16102 16102 IN IP4 10.2.11.35..s=s ession..c=IN IP4 10.2.11.35..t=0 0..m=audio 14640 RTP/AVP 0 8 4 111 18 3 97 7 110 5 101..a=rtpmap:0 PCMU/8000..a=rtpmap :8 PCMA/8000..a=rtpmap:4 G723/8000..a=rtpmap:111 G726-32/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:3 G SM/8000..a=rtpmap:97 iLBC/8000..a=rtpmap:7 LPC/8000..a=rtpmap:110 speex/8000..a=rtpmap:5 DVI4/8000..a=rtpmap:101 teleph one-event/8000..a=fmtp:101 0-16....

...stion 2: Is there a config option to force * to just passthrough the codec list sent by the 7960 in the invite? Question 3: What are SDP codec matching rules for SIP endpoints? How do they decide on common codec. Comparing the SDP sent and receive all systems claim support for 3 common codecs: a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 Now of course each device specified these 3 codecs in a different order. Under normal circumstances I feel this call should complete why is * claiming a codec mismatch? - Dustin - From phone v=0 o=Cisco-SIPUA 5892 12461 IN IP4 192.168.68...

...ip.conf I have: disallow=all ; 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=160...

...is through). Both the client?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...

...:111111111111@192.0.0.0> Call-ID: 265fdf1e0f61bd2a3422940c4ec90131@192.0.0.0 CSeq: 102 INVITE User-Agent: Asterisk PBX Content-Type: application/sdp Content-Length: 384 v=0 o=root 21604 21604 IN IP4 192.0.0.0 s=session c=IN IP4 192.0.0.0 t=0 0 m=audio 27916 RTP/AVP 4 3 18 0 8 14 5 10 7 110 97 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:14 MPA/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:110 SPEEX/8000 a=rtpmap:97 iLBC/8000 (no NAT) to 66.33.146.12:5060 -- Called 1800XXXXXXX@net2phone Sip rea...

...upported: replaces Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 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:...

...-416B-92B7-DD24DA4630E2@192.168.11.9. CSeq: 4807 INVITE. Max-Forwards: 69. Content-Type: application/sdp. User-Agent: X-LITE build 1082. Content-Length: 321. . v=0. o=101 45727796 45727796 IN IP4 192.168.11.9. s=X-LITE. c=IN IP4 my.openser.ip.addr. t=0 0. m=audio 35066 RTP/AVP 0 8 3 18 98 97 101. a=rtpmap:0 pcmu/8000. a=rtpmap:8 pcma/8000. a=rtpmap:3 gsm/8000. a=rtpmap:18 G729/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:97 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 2007/05/17 13:31:36.325713 my.openser.ip.addr:5060 -> my.asterisk.ip.addr :5060 INVITE sip:03939749001@my.asteris...

...arted. When debugging 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=...

...E-D94F-060D-2F43-D9557F056D54 at 10.44.1.20 CSeq: 41181 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 304 v=0 o=102 3079422269 3079422292 IN IP4 10.44.1.20 s=X-Lite c=IN IP4 10.44.1.20 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Attempting SIP protocol listen on: 10.44.1.20:5060 Established SIP protocol listen on: 10.44.1.20:5060 SEND TIME: 3079423989 S...

...e:Hc1e19c54 2 UDP 2130706430 2607:abcd:9876:5432::247 14385 typ host a=candidate:Hf5a91683 2 UDP 2130706430 2001:1234:5678:abcd::2 14385 typ host a=candidate:Hdccea0f2 2 UDP 2130706430 2001:123:ab:123::2 14385 typ host a=candidate:Hcbb5ed22 2 UDP 2130706430 fe80::21f:c6ff:fec4:926a 14385 typ host a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:118 L16/16000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:117 spee...

...ed CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Fri, 04 Jun 2004 02:26:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 461 v=0 o=root 284 284 IN IP4 munged s=session c=IN IP4 munged t=0 0 m=audio 14466 RTP/AVP 0 8 4 3 2 5 10 7 18 110 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=rtpmap:110 SPEEX/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - -...

...VITE User-Agent: sipgate asterisk Date: Mon, 10 Oct 2005 15:53:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 448 v=0 o=root 5903 5903 IN IP4 217.10.79.218 s=session c=IN IP4 217.10.79.55 t=0 0 m=audio 44214 RTP/AVP 8 0 3 97 18 2 4 5 110 7 10 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:110 speex/8000 a=rtpmap:7 LPC/8000 a=rtpmap:10 L16/8000 a=silenceSupp:off - - - - a=direction:active a=nortpproxy:yes 17 heade...

Hi, I have 2 systems (A and B). I have an 800 number... when someone calls the 800 number it goes: IAX2-->A---IAX---B--->SIP PHONE However.. if the user calling the 800 number is a SIP user that is registered to A it goes: SIP--->A---IAX---B--->SIP PHONE This is the problem... when a call comes in from the IAX2 800 provider, things work fine... however if a SIP user registered to

SDP question if * recieves "a=rtpmap:103 telephone-event/8000" it shouldn't it send out the same "a=rtpmap:103 telephone-event/8000" to the other side of the connection? and not something like "a=rtpmap:101 telephone-event/8000"? Thanks

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

...<sip:asterisk@192.168.1.100> Call-ID: 2a3fe9bc1d8dd93a400263c775c63f5b@192.168.1.100 CSeq: 102 INVITE User-Agent: Asterisk PBX Content-Type: application/sdp Content-Length: 236 v=0 o=root 3296 3296 IN IP4 192.168.1.100 s=session c=IN IP4 192.168.1.100 t=0 0 m=audio 10860 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 (no NAT) to 213.137.73.140:5060 -- Called 16507148980@iconnect Retransmitting #1 (no NAT): INVITE sip:16507148980@natrelay.deltathree.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9h...

...d622d6836df2d7a7c2c21b39b14@200.61.32.238 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 261 v=0 o=root 20500 20500 IN IP4 200.61.32.238 s=session c=IN IP4 200.61.32.238 t=0 0 m=audio 15740 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 m=video 12476 RTP/AVP Oct 22 09:19:49.938: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 200.61.32.238:5060;branch=z9hG4bK67299947 From: "52880472" <sip:52880472@200.61.32.238>;tag=as...

...ITE User-Agent: Grandstream SIP UA 1.0.3.81 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Content-Length: 257 v=0 o=btel 0 0 IN IP4 192.168.1.21 s=- c=IN IP4 192.168.1.21 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 12 headers, 13 lines Using latest request as basis request Sending to 192.168.1.21 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format ULAW Foun...

...ons: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:126 telephone-event/8000 a=maxptime:60 a=ssrc:2179369454 cname:SvzCJjIAujxHGm9P a=ssrc:217936945...

...TU-T G.723 Media Format: ITU-T G.711 PCMA Media Format: ITU-T G.729 Media Format: 96 Media Format: 97 Media Format: 98 Media Format: 100 Media Format: 101 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute Fieldname: rtpmap Media Attribute Value: 0 PCMU/8000 Media Attribute (a): rtpmap:2 G726-32/8000 Media Attribute Fieldname: rtpmap Media Attribute Value: 2 G726-32/8000 Media Attribute...