RTP support in Burrito
Hi,
Do burrito support RTP/RTSP . I need to build a rtp streaming app for android. Can Burrito be used for it ?
If possible pls give some reference link .
Regards,
Amit
There is no RTP support in J2ME. Vikram Goyal has done some experiments using current mmapi but it does not work because of limitations of Manager and Player classes.
http://today.java.net/pub/a/today/2006/08/22/experiments-in-streaming-java-me.html?page=1
Similar Messages
-
Darshan Prajapati -- About RTP support in J2ME
Hello friends I am developing the SIP client on J2ME. I want to route speech packets in real time in both client server direction. Also I want to continuously monitor the microphone to get the speech of the user and then simultaneously encode that speech and send it over wifi or internet through RTP packets.
So how can I do this? Please tell me api for getting continuous input from microphone(I do not want to first record and then send it.) and also making RTP packets to send it on internet.
Thanks in advance !
Darshan PrajapatiThere is no RTP support in J2ME. Vikram Goyal has done some experiments using current mmapi but it does not work because of limitations of Manager and Player classes.
http://today.java.net/pub/a/today/2006/08/22/experiments-in-streaming-java-me.html?page=1 -
hi,
i am doing a proj to stream captured video to mobile phones. i can stream them using rtp. but in the mobile, i am not able to receive it.
the code is
player = Manager.createPlayer("rtp://192.168.10.81:42060/video");
this throws a mediaexception: cannot create player for rtp 192.168.10.81:42060/video
can anybody help me plz... its urgent.
thanks in advance
premithaHI all,...
appearantly we all have the same problems...
i want 2 implement rtp player for ipaq h5550. Ive used j9 but i dont know how to implement mmapi to ipaq so j9 can run it....
i want 2 use the jeode vm with jmf, but i dont have any jeode coz ive lost my cd...and they (HP) doesnt support it anymore,
i heard in the forum that rtp player worked with j9 and jmf crossplatform. so would anyone who have made it please help all of us,..
im beginning 2 felt sick about this.....
thx
regards ,
hendra -
Hi All,
I want to add support in jmf to stream RTP streams for MP4 files. For this i used Fobs4Jmf to read mp4 files from the local system. As fobs4jmf only read MP4 file from local system and render it on jmstudio, but not able to create MP4V/RTP and MP4A/RTP, therefore i tried writing my own packetiser for MP4 files. But then i realised fobs4jmf uses FFMPEG_VIDEO codec to read all videos and FFMPEG_AUDIO codec to read all audios. FFMPEG_VIDEO converts all video types MP4,mpeg etc to RGB format and FFMPEG_Audio converts all audio types to LINEAR format. Now i want to send this data in MP4/RTP packet but have no idea how to convert RGB in MP4V and Linear into MP4A.
I think i am missing something. Or i am doing it all together wrong.
My ultimate aim is to stream videos to mobile using RTSP. I have written my RTSP server but for that i need RTP streams.
Is there any other way to stream RTP streams to mobile from jmf.
Kindly help me i am totally stuck.search4rajat wrote:
Hi, is anyone there who can help me. Where is captfoss and other gurus of jmf. I was on vacation.
Kindly help me it is very important. Any slight insight may be helpful.... like how to stream RTP to mobile.I doubt it's "very important"...
If the remote system handles MP4/RTP streams, then you'll likely need to do nothing else but write an MP4 packetizer... and the packetizer probably won't need to do anything but break the file up into peices, no special processing.
[http://java.sun.com/javase/technologies/desktop/media/jmf/2.1.1/solutions/CustomPayload.html]
Modify that example so that instead of taking in AudioFormat.LINEAR and outputting CUSTOM_PCM, make it so it takes in MP4 and outputs MP4_RTP... and just play around with the example. I've never written a custom packetizer, so I don't have any grand advice other than, use that example and try to create a simple MP4 packetizer.
Your processor shouldn't be transcoding from MP4 to RGB if you're wanting to transmit in MP4-RTP. It should be transcoding from MP4 directly to MP4-RTP, which is what your custom packetizer will do... -
I want to connect mobile device with a remote PC that has created an RTP session .How can I do this ?
I checked with this
Manager.createPlayer("rtp://127.0.0.1:9998/video");
but it is not working and an exception is thrown
so please help me if any one has idea abot this?.For my project, I initially considered JMF...but ran into issues due to the level of abstraction it provides. With JMF, I was able to play audio easily but to actually do anything else with it was almost impossible or very difficult given the provided interfaces. And judging from the last release, I don't think Sun is actively supporting it.
My project plays live streaming audio over RTP using Java Sound. There is a Java library that implements the RTP protocol called jrtp which can be found at https://jrtp.dev.java.net. -
Re: Dubt about the JMF 2.1.1 - Supported Formats and RTP supported
Post your JMF question on that forum:
http://forum.java.sun.com/forum.jspa?forumID=28It can be done in Designer, but I don't think it can be done with AcroForms (forms within Acrobat).
I should mention that there is an option that allows the latter. In that case you have to convert the DOC file to an HTML file in WORD (it should be a fillable HTML). Then use create PDF from web page to make the fillable form in Acrobat. -
Hi,
I am trying out a "live" application in FMS3.0 which
involoves streaming a live feed from a host in rtp format to
fms3,and i want fms to broadcast this live feed again...so just
wanted to know whether anyone has tried rtp support for
fms3.FMS3 doesn't support RTP/RTSP stream ingest.
-
I am looking for the download page for JMF 2.1.1 implementation.
I have looked in the following places with no success.
JMF Setup
http://java.sun.com/products/java-media/jmf/2.1.1/setup.html
http://java.sun.com/products/java-media/jmf/2.1.1/setup-java.html
http://java.sun.com/products/java-media/jmf/2.1.1/requirements.html#win
Voice Chat
http://java.sun.com/products/java-media/jmf/2.1.1/solutions/
http://java.sun.com/products/java-media/jmf/2.1.1/solutions/ToolsTx.html
RTP Support
http://java.sun.com/products/java-media/jmf/2.1.1/support-rtp.html
Please help.
Thanks.http://www.sun.com/software/communitysource/jmf/download.xml
Yes implementation, or as Sun calls it a Reference Implementation.
wew -
live-media is "A set of C++ libraries for multimedia streaming" especially usefull with RTSP Servers.
As I was going to compile mplayer with stream support on Arch 64, I noticed that live-media in /extra is outdated : 2006.03.03 when 2006.10.12a is available.
New version supports stuff like H264VideoRTPSink for transmitting H.264/RTP streams, updated H364VideoRTPSink, MPEG2TransportStreamFromESSource.
Here a proposal for an updated PKGBUILD:
# Maintainer : Aaron, phrakture, Griffin <aaron>
# Contributor: Gilles CHAUVIN <gcnweb>
pkgname=live-media
pkgver=2006.10.12a
pkgrel=1
pkgdesc="A set of C++ libraries for multimedia streaming"
arch=(i686 x86_64)
url="http://live555.com/liveMedia/"
depends=('gcc')
source=(http://live555.com/liveMedia/public/live.$pkgver.tar.gz)
md5sums=('5d6578fa37af15b1c85edf65b0f23a48')
(as I changed only the version & md5sum, I kept Contributor as it was)
It compiled & installed fine, namcap says nothing. dunno why mplayer complains about "missing" 'liveMedia.hh' & other '.hh' libs as they _are_ in /usr/lib/live-media/ !This live-media allows to listen rtp/rtsp stream with vlc or mplayer. Question is how to create such a stream with arch box? Like now I use mpd and icecast to do http streaming, which can be only listened from one source at the time due to different delays.
I would like to have synchronized audio from 3 boxes running mplayers in my house. Mplayer is required as xbox uses mplayer. I cannot find a player and streamer combination that could do rtp/rtsp stream.
I could not get this to work http://www.live555.com/mediaServer
Apple darwin streaming server looks a bit difficult to build ( http://www.streamingmedia.com/tutorials … ial_id=143 ).
Flumotion did not seem to have rtp support included yet or does gstreamers rtp support work?
Suggestions?
Last edited by Purch (2007-03-05 17:37:05) -
SIP CUBE sees DTMF * and # parsed as the number 1
I am integrating a third party ippbx to Cisco Cube all working but with the UCCX, because when I send the * or # dtmf digits the cube sees them as the number 1 dtmf digit
I have verified with wireshark that vendor system is sending correctly but cube reads/interprets it as the number 1 any help would be appreciatedIf you can see the attached, I dialed 1234#
and the cube sent 12341
AAPUSVAARPTVG02#
009451: Mar 4 12:51:20.798: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIDeleteContextFromTable: Context for key=[201] removed.
009452: Mar 4 12:51:20.798: //32189/48265AD0BE6D/SIP/Info/sipSPIUdeleteCcbFromUASReqTable: ****Deleting from UAS Request table.
009453: Mar 4 12:51:20.798: //32189/48265AD0BE6D/SIP/Info/sipSPIUdeleteCcbFromTable: Deleting from table. ccb=0x4A0D64B0 [email protected]
009454: Mar 4 12:51:20.798: //32189/48265AD0BE6D/SIP/Info/sipSPIUdeleteCcbFromUASRespTable: ****Deleting from UAS Response table.
009455: Mar 4 12:51:20.798: //32189/48265AD0BE6D/SIP/Info/sipSPIUdeleteCcbFromTable: Deleting from table. ccb=0x4A0D64B0 [email protected]
009456: Mar 4 12:51:20.798: //32189/48265AD0BE6D/SIP/Info/sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd
009457: Mar 4 12:51:20.798: //32189/48265AD0BE6D/SIP/Info/ccsip_qos_cleanup: Entry
009458: Mar 4 12:51:20.798: //32189/48265AD0BE6D/SIP/Info/sipSPI_ipip_free_codec_profile: Codec Profiles Freed
009459: Mar 4 12:51:20.798: //32189/48265AD0BE6D/SIP/Info/sipSPIUfreeOneCCB: Freeing ccb 4A0D64B0
AAPUSVAARPTVG02#
AAPUSVAARPTVG02#
AAPUSVAARPTVG02#
AAPUSVAARPTVG02#
AAPUSVAARPTVG02#
AAPUSVAARPTVG02#
AAPUSVAARPTVG02#
AAPUSVAARPTVG02#
AAPUSVAARPTVG02#
AAPUSVAARPTVG02#
AAPUSVAARPTVG02#
AAPUSVAARPTVG02#
AAPUSVAARPTVG02#
AAPUSVAARPTVG02#
AAPUSVAARPTVG02#
AAPUSVAARPTVG02#
AAPUSVAARPTVG02#
AAPUSVAARPTVG02#
AAPUSVAARPTVG02#
AAPUSVAARPTVG02#
AAPUSVAARPTVG02#
AAPUSVAARPTVG02#
AAPUSVAARPTVG02#
009460: Mar 4 12:51:54.243: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [10.107.177.95]:5060
009461: Mar 4 12:51:54.243: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1
009462: Mar 4 12:51:54.243: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000
009463: Mar 4 12:51:54.243: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.107.177.95;branch=z9hG4bK+ccab88001db7a774e18a6626099dfafe+10.107.177.95+8
Allow-Events: refer
Allow-Events: message-summary
Allow-Events: dialog
Max-Forwards: 70
Call-ID: [email protected]
From: "Manager";tag=10.107.177.95+8+1740001+dc7af6ae
To: ""
CSeq: 392882336 INVITE
Expires: 180
Supported: replaces
Contact: "Manager"
Content-Type: application/sdp
Content-Length: 195
User-Agent: Wave/10.5.3021.2152
v=0
o=InstantOffice 785 0 IN IP4 10.107.177.95
s=phone-call
c=IN IP4 10.107.177.95
t=0 0
m=audio 16746 RTP/AVP 0 18
a=rtpmap:0 pcmu/8000/1
a=rtpmap:18 g729/8000/1
a=ptime:20
a=sendrecv
009464: Mar 4 12:51:54.243: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
009465: Mar 4 12:51:54.247: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAddContextToTable: Added context(0x4A0E0A28) with key=[203] to table
009466: Mar 4 12:51:54.247: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 10.107.177.95,Port 5060, Transport 1, SentBy Port 5060
009467: Mar 4 12:51:54.247: //-1/742AD833BE83/SIP/State/sipSPIChangeState: 0x4A0E0A28 : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)
009468: Mar 4 12:51:54.247: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 10.107.177.95,Port 5060, Transport 1, SentBy Port 5060
009469: Mar 4 12:51:54.247: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone EST to SIP default timezone = GMT
009470: Mar 4 12:51:54.247: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 10.107.177.95,Port 5060, Transport 1, SentBy Port 5060
009471: Mar 4 12:51:54.247: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICheckIpip: VOIP dialpeer (peer=0x474A30BC) found for sip_user: 4095
009472: Mar 4 12:51:54.247: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentGTD: No GTD found in inbound container
009473: Mar 4 12:51:54.247: //-1/742AD833BE83/SIP/Info/sipSPIUaddCcbToUASReqTable: ****Adding to UAS Request table.
009474: Mar 4 12:51:54.247: //-1/742AD833BE83/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x4A0E0A28 [email protected]
009475: Mar 4 12:51:54.247: //-1/742AD833BE83/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on carrier id
009476: Mar 4 12:51:54.247: //-1/742AD833BE83/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on Incoming called number: 4095
009477: Mar 4 12:51:54.247: //-1/742AD833BE83/SIP/Info/sipSPIMatchSrcIpGroup: Match not found on destination pattern: 100
009478: Mar 4 12:51:54.247: //-1/742AD833BE83/SIP/Info/ccsipUpdateIncomingCallParams: ccCallInfo: Calling name Manager, number 100, Calling oct3 0x00, oct_3a 0x80, Called number 4095
009479: Mar 4 12:51:54.247: //-1/742AD833BE83/SIP/Info/sipSPIGetShrlPeer: Try match incoming dialpeer for Calling number: : 100
009480: Mar 4 12:51:54.251: //-1/742AD833BE83/SIP/Info/sipSPIGetCallConfig: Precondition tag absent in Require/Supported header
009481: Mar 4 12:51:54.251: //-1/742AD833BE83/SIP/Info/sipSPIGetCallConfig: Non dial peer leg - using RTP Supported Codecs
009482: Mar 4 12:51:54.251: //-1/742AD833BE83/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 18
009483: Mar 4 12:51:54.251: //-1/742AD833BE83/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 0
009484: Mar 4 12:51:54.251: //-1/742AD833BE83/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 8
009485: Mar 4 12:51:54.251: //-1/742AD833BE83/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 9
009486: Mar 4 12:51:54.251: //-1/742AD833BE83/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 4
009487: Mar 4 12:51:54.251: //-1/742AD833BE83/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 2
009488: Mar 4 12:51:54.251: //-1/742AD833BE83/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 15
009489: Mar 4 12:51:54.251: //-1/742AD833BE83/SIP/Info/sipSPIGetCallConfig: RTP Preferred Codecs supported by GW 3
009490: Mar 4 12:51:54.251: //-1/742AD833BE83/SIP/Info/sipSPIContinueNewMsgInvite: Calling name Manager, number 100, Calling oct3 0x00, oct_3a 0x80, ext_priv 0x00, Called number 4095, oct3 0x00
009491: Mar 4 12:51:54.251: //-1/742AD833BE83/SIP/Info/sipSPIContinueNewMsgInvite: Carrier id code , prev_cid NONE, next_cid NONE, prev_tgrp NONE, next_tgrp NONE
009492: Mar 4 12:51:54.251: //-1/742AD833BE83/SIP/Info/sipSPIRscmsmAvail: Value returned by check is = 0
009493: Mar 4 12:51:54.251: //32208/742AD833BE83/SIP/Info/sipSPIProcessHistoryInfoHeader: No HI headers recvd from app container
009494: Mar 4 12:51:54.251: //32208/742AD833BE83/SIP/Info/sipSPIProcessDiversionHeader: No diversion headers recvd from app container
009495: Mar 4 12:51:54.251: //32208/742AD833BE83/SIP/Info/sipSPIProcessReplacesHeader: No replaces hdr found
009496: Mar 4 12:51:54.251: //32208/742AD833BE83/SIP/Info/sipSPIDoMediaNegotiation: Number of m-lines = 1
SIP: (32208) Attribute mid, level 1 instance 1 not found.
009497: Mar 4 12:51:54.251: //32208/742AD833BE83/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 172.25.30.21
009498: Mar 4 12:51:54.251: //32208/742AD833BE83/SIP/Info/sipSPIDoAudioNegotiation: Codec (g711ulaw) Negotiation Successful on Static Payload for m-line 1
009499: Mar 4 12:51:54.251: //32208/742AD833BE83/SIP/Info/sipSPIDoPtimeNegotiation: One ptime attribute found - value:20
009500: Mar 4 12:51:54.251: //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values :Codec: g711ulaw ptime :20, codecbytes: 160
009501: Mar 4 12:51:54.251: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711ulaw codecbytes :160, ptime: 20
009502: Mar 4 12:51:54.251: //32208/742AD833BE83/SIP/Media/sipSPIDoPtimeNegotiation: Offered ptime:20, Negotiated ptime:20 Negotiated codec bytes: 160 for codec g711ulaw
009503: Mar 4 12:51:54.251: //32208/742AD833BE83/SIP/Info/sipSPIDoDTMFRelayNegotiation: m-line index 1
009504: Mar 4 12:51:54.251: //32208/742AD833BE83/SIP/Info/sipSPIDoDTMFRelayNegotiation: Requested DTMF-RELAY option(s) not found in Preferred DTMF-RELAY option list!
009505: Mar 4 12:51:54.251: //32208/742AD833BE83/SIP/Info/sipSPIStreamTypeAndDtmfRelay: DTMF Relay mode: Inband Voice
009506: Mar 4 12:51:54.251: //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: NSE payload from X-cap = 0
009507: Mar 4 12:51:54.251: //32208/742AD833BE83/SIP/Info/sip_select_modem_relay_params: X-tmr not present in SDP. Disable modem relay
009508: Mar 4 12:51:54.251: //32208/742AD833BE83/SIP/Info/sipSPIGetSDPDirectionAttribute: No direction attribute present or multiple direction attributes that can't be handled for m-line:1 and num-a-lines:0
009509: Mar 4 12:51:54.251: //32208/742AD833BE83/SIP/Info/sipSPIDoAudioNegotiation: Codec negotiation successful for media line 1
payload_type=0, codec_bytes=160, codec=g711ulaw, dtmf_relay=inband-voice
stream_type=voice-only (0), dest_ip_address=10.107.177.95, dest_port=16746
009510: Mar 4 12:51:54.251: //32208/742AD833BE83/SIP/State/sipSPIChangeStreamState: Stream (callid = -1) State changed from (STREAM_DEAD) to (STREAM_ADDING)
009511: Mar 4 12:51:54.251: //32208/742AD833BE83/SIP/Media/sipSPIUpdCallWithSdpInfo:
Preferred Codec : g729r8, bytes :20
Preferred DTMF relay : inband-voice
Preferred NTE payload : 101
Early Media : No
Delayed Media : No
Bridge Done : No
New Media : No
DSP DNLD Reqd : No
009512: Mar 4 12:51:54.251: //32208/742AD833BE83/SIP/Info/resolve_media_ip_address_to_bind: Media already bound, use existing source_media_ip_addr
009513: Mar 4 12:51:54.251: //32208/742AD833BE83/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 172.25.30.21
009514: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_report_media_to_peer:
callId 32208 peer 0 flags 0x201 state STATE_IDLE
009515: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
CallID 32208, sdp 0x4A606AE0 channels 0x4A0E1CC8
009516: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/copy_channels:
callId 32208 size 0 ptr 0x495086DC)
009517: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 0 mline 1
009518: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting codec g711ulaw
009519: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/codec_found:
Codec to be matched: 5
009520: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: ADD AUDIO CODEC 5
009521: Mar 4 12:51:54.255: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711ulaw codecbytes :160, ptime: 20
009522: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Media negotiation done: stream->negotiated_ptime=20,stream->negotiated_codec_bytes=160, coverted ptime=20 stream->mline_index=1, media_ndx=1
009523: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Error/sipSPI_ipip_copy_sdp_to_channelInfo:
failed to update call entry
009524: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Adding codec 5 ptype 0 time 20, bytes 160 as channel 0 mline 1 ss 1 10.107.177.95:16746
009525: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 18 mline 1
009526: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Media/sipSPISelectCodecVersion: Codec (g729br8) is not in preferred list
009527: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: An exact codec match not configured, using interoperable codec g729r8
009528: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting codec g729r8
009529: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/codec_found:
Codec to be matched: 16
009530: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: ADD AUDIO CODEC 16
009531: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Media negotiation NOT done, get ptime from sdp: ptime=20, media_ndx=1
009532: Mar 4 12:51:54.255: //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values :Codec: g729r8 ptime :20, codecbytes: 20
009533: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Error/sipSPI_ipip_copy_sdp_to_channelInfo:
failed to update call entry
009534: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Adding codec 16 ptype 18 time 20, bytes 20 as channel 1 mline 1 ss 1 10.107.177.95:16746
009535: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Copy sdp to channel- AFTER CODEC FILTERING: ccb->pld.ipip_caps.codecInfo[channel_ndx].codec = 5
009536: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Copy sdp to channel- AFTER CODEC FILTERING: ccb->pld.ipip_caps.codecInfo[channel_ndx].codec = 16
009537: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Copy sdp to channel- AFTER CODEC FILTERING: ccb->pld.ipip_caps.codecInfo[channel_ndx].codec = -1
009538: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_report_media_to_peer:
callId 32208 flags 0x100 state STATE_IDLE
009539: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_report_media_to_peer:
Report initial call media
009540: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_report_media_to_peer: ccb->flags 0xC, ccb->pld.flags_ipip 0x201
009541: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/copy_channels:
callId 32208 size 468 ptr 0x47E3CD94)
009542: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_report_media_to_peer:
CCSIP: Unable to report channel ind
009543: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/ccsip_update_srtp_caps: 5054: Posting Remote SRTP caps to other callleg.
009544: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_report_media_to_peer: do cc_api_caps_ind()
009545: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Media/sipSPIUpdCallWithSdpInfo:
Stream type : voice-only
Media line : 1
State : STREAM_ADDING (2)
Stream address type : 1
Callid : -1
Negotiated Codec : g711ulaw, bytes :160
Nego. Codec payload : 0 (tx), 0 (rx)
Negotiated DTMF relay : inband-voice
Negotiated NTE payload : 0 (tx), 0 (rx)
Negotiated CN payload : 0
Media Srce Addr/Port : [172.25.30.21]:0
Media Dest Addr/Port : [10.107.177.95]:16746
009546: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPIHandleInviteMedia:
Negotiated Codec : g711ulaw, bytes :160
Preferred Codec : g729r8, bytes :20
Preferred DTMF relay 1 : 0
Preferred DTMF relay 2 : 0
Negotiated DTMF relay : 0
Preferred and Negotiated NTE payloads: 101 0
Preferred and Negotiated NSE payloads: 100 0
Preferred and Negotiated Modem Relay: 0 0
Preferred and Negotiated Modem Relay GwXid: 1 0
009547: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPIDoQoSNegotiationWithMediaLine: Entry
009548: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPIDoQoSNegotiationWithMediaLine: QOS negotiation for mline_index 1
009549: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPIDoStreamQoSNegotiation: Best effort
009550: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPICanSetFallbackFlag: Local Fallback is not active
009551: Mar 4 12:51:54.255: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 18716 for stream 1
009552: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPIUpdateSrcSdpFixedPart: Reserving rtp port for stream 1, src_port=18716
009553: Mar 4 12:51:54.255: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetMediaDirectionForStream: Setting Media direction SENDRECV for stream 1
009554: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPIUpdateSrcSdpVariablePart: Setting stream 1 portnum to 18716
009555: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPIUpdateSrcSdpVariablePart:
SIP update src sdp, negoitated codec 5, payload type 0
009556: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPIAddBillingInfoToCcb: sipCallId for billing records = [email protected]
009557: Mar 4 12:51:54.255: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentCPA: No CPA found in inbound container
009558: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPIProcessCPA: No x-cisco-cpa content found
009559: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_store_channel_info: Store channelInfo in CallInfo
009560: Mar 4 12:51:54.255: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_store_channel_info: dtmf negotiation done, storing negotiated dtmf = 0,
009561: Mar 4 12:51:54.259: //32208/742AD833BE83/SIP/Info/sipSPIShrlCall: Check peer: 0 for Shared-Line call, callid: 32208
009562: Mar 4 12:51:54.259: //32208/742AD833BE83/SIP/Info/ccsip_set_bearer_capability:
Bearer Capability: Speech (0x00)
009563: Mar 4 12:51:54.259: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQSIG: No QSIG Body found in inbound container
009564: Mar 4 12:51:54.259: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQ931: No RawMsg Body found in inbound container
009565: Mar 4 12:51:54.259: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICreateNewRawMsg: No Data to form The Raw Message
009566: Mar 4 12:51:54.259: //32208/742AD833BE83/SIP/Info/sipSPIContinueNewMsgInvite: ccsip_api_call_setup_ind returned: SIP_SUCCESS
009567: Mar 4 12:51:54.259: //32208/742AD833BE83/SIP/Info/sipSPIUaddccCallIdToTable: Adding call id 7DD0 to table
009568: Mar 4 12:51:54.259: //32208/742AD833BE83/SIP/Transport/sipSPITransportSendMessage: msg=0x4A3DEDB8, addr=10.107.177.95, port=5060, sentBy_port=5060, is_req=0, transport=1, switch=0, callBack=0x00000000
009569: Mar 4 12:51:54.259: //32208/742AD833BE83/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
009570: Mar 4 12:51:54.259: //32208/742AD833BE83/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
009571: Mar 4 12:51:54.259: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x4A3DEDB8, addr=10.107.177.95, port=5060, connId=0 for UDP
009572: Mar 4 12:51:54.259: //32208/742AD833BE83/SIP/State/sipSPIChangeState: 0x4A0E0A28 : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_RECD_INVITE, SUBSTATE_NONE)
009573: Mar 4 12:51:54.259: //32208/742AD833BE83/SIP/Info/sipSPIProcessContactInfo: Previous Hop 10.107.177.95:5060
009574: Mar 4 12:51:54.259: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_PROCEEDING
009575: Mar 4 12:51:54.263: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAddContextToTable: Added context(0x4A0D11F4) with key=[204] to table
009576: Mar 4 12:51:54.263: //32209/000000000000/SIP/State/sipSPIChangeState: 0x4A0D11F4 : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)
009577: Mar 4 12:51:54.263: //32209/000000000000/SIP/Info/ccsip_call_setup_request: Before processing SETUP REQccb->pld.flags_ipip = 200
009578: Mar 4 12:51:54.263: //32209/000000000000/SIP/Info/ccsip_call_setup_request: midcall-signaling passthru enabled
009579: Mar 4 12:51:54.263: //32209/000000000000/SIP/Info/ccsip_call_setup_request:
This a IPIP call: Chan 0, codec 5 channel 16746, ip 10.107.177.95:16746 params 0x4A439388 caps 0x47E60E64
009580: Mar 4 12:51:54.263: //32209/000000000000/SIP/Info/ccsip_call_setup_request:
This a IPIP call: Chan 1, codec 16 channel 16746, ip 10.107.177.95:16746 params 0x4A439388 caps 0x47E60E64
009581: Mar 4 12:51:54.263: //32209/000000000000/SIP/Info/ccsip_gw_set_sipspi_mode: Setting SPI mode to SIP-SIP
009582: Mar 4 12:51:54.263: //32209/000000000000/SIP/Info/ccsip_call_setup_request: After processing SETUP REQccb->pld.flags_ipip = 400000
009583: Mar 4 12:51:54.263: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetOutboundHostAndDestHostPrivate: CCSIP: target_host : 172.24.7.14 target_port : 5060
009584: Mar 4 12:51:54.263: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP
009585: Mar 4 12:51:54.263: //32209/742AD833BE83/SIP/Info/ccsip_call_setup_request: Incrementing call counter in dial-peer [11]
009586: Mar 4 12:51:54.263: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler:
009587: Mar 4 12:51:54.263: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler: switch(ev.ev_id: 162)
009588: Mar 4 12:51:54.263: //32208/742AD833BE83/SIP/Info/ccsip_event_handler:
ccsip_event_handler: peer ID 32209 chans 0x4A6AF8B8 event 162 flags 0x40001C 0x100 0x601 data 0x4A6AF8B8
009589: Mar 4 12:51:54.263: //32208/742AD833BE83/SIP/Info/ccsip_event_handler:
ccsip_event_handler: CC_EV_H245_SET_MODE: peer ID 32209 chans 0x4A6AF8B8 event 162 flags 0x40001C 0x100 0x601 data 0x4A6AF8B8, type = 1
009590: Mar 4 12:51:54.263: //32208/742AD833BE83/SIP/Info/ccsip_gw_set_sipspi_mode: Setting SPI mode to SIP-SIP
009591: Mar 4 12:51:54.263: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler: CC_R_SUCCESS_WITH_CONFIRMED
009592: Mar 4 12:51:54.267: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 3
009593: Mar 4 12:51:54.267: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 2
009594: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Info/sipSPIUaddccCallIdToTable: Adding call id 7DD1 to table
009595: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Info/sipSPIGetCallConfig: preferred_codec set[0] type :No Codec bytes: 0
009596: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Info/sipSPIGetCallConfig: Media forking disabled
009597: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Info/sipSPICanSetFallbackFlag: Local Fallback is not active
009598: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Info/sipSPIGetCallConfig: Using Voice Class Codec, tag = 1
009599: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Media/sipSPICopyPeerDataToCCB: Firewall traversal is not enabled
009600: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Info/sipSPIGetCallConfig: xcoder high-density disabled
009601: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Info/sipSPIGetCallConfig: Flow Mode set to FLOW_THROUGH
009602: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Info/sipSPIGetCallConfig: Media forking disabled
009603: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Info/preprocessSetup:
This is a not a SIGO Call -, could be DM call
009604: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_call_setup: No video caps posted by peer
009605: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_call_setup: xcoder high-density disabled
009606: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_call_setup: Flow Mode set to FLOW_THROUGH
009607: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_copy_channelInfo_to_sdp:
callid 32209, channels 0x47DBA6E4 caps 0x47E60E64
009608: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_copy_channelInfo_to_sdp: Peer cap provided: callid = 32209, peer dtmf = 0
009609: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_copy_channelInfo_to_sdp: callid = 32209, peer not doing RFC2833, peer dtmf = 0, enable NTE_ASSUMED
009610: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Info/codec_found:
Codec to be matched: 5
009611: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Info/codec_found:
Codec to be matched: 16
009612: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Info/sipSPIDtmfTranscoder: Return upon SCCP version 0
009613: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Info/codec_found:
Codec to be matched: 5
009614: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Info/codec_found:
Codec to be matched: 16
009615: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Info/sipSPIDtmfTranscoder: Return upon SCCP version 0
009616: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Info/sipSPIValidateAndCopyOutboundHost: CCSIP: copy target_host to outbound_host
009617: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 172.25.30.21
009618: Mar 4 12:51:54.267: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 16534 for stream 1
009619: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Media/sipSPIAddSDPMediaPayload: Preferred method of dtmf relay is: 6, with payload: 101
009620: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Info/sipSPIAddSDPPayloadAttributes:
max_event 15
009621: Mar 4 12:51:54.267: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711ulaw codecbytes :160, ptime: 20
009622: Mar 4 12:51:54.267: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetMediaDirectionForStream: Setting Media direction SENDRECV for stream 1
009623: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Info/sip_generate_sdp_xcaps_list: Modem Relay and T38 disabled. X-cap not needed
009624: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Info/sipSPIOutgoingCallSDP: Creating recv-only stream for outbound call
009625: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Media/sipSPIProcessRtpSessions: sipSPIProcessRtpSessions
009626: Mar 4 12:51:54.267: //32209/742AD833BE83/SIP/Media/sipSPIProcessRtpSessions: No active streams.
009627: Mar 4 12:51:54.271: //32209/742AD833BE83/SIP/Info/sip_gw_pre_setup_add_sdp_container: SDP container added
009628: Mar 4 12:51:54.271: //32209/742AD833BE83/SIP/Info/sipSPIAddMLPPServicesInfo: No MLP Info available on incoming leg
009629: Mar 4 12:51:54.271: //32209/742AD833BE83/SIP/Info/sipSPIShrlGetInstanceInfo: Obtained the call instance 0 for non-shared-line '....' with callid: 32209
009630: Mar 4 12:51:54.271: //32209/742AD833BE83/SIP/Info/sipSPIAddCiscoGcid: Gcid value not set - not adding header.
009631: Mar 4 12:51:54.271: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_set_history_info_header: No HI header recvd from container
009632: Mar 4 12:51:54.271: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_set_diversion_header: No diversion header recvd from container
009633: Mar 4 12:51:54.271: //32209/742AD833BE83/SIP/Info/act_idle_continue_call_setup:
009634: Mar 4 12:51:54.271: //32209/742AD833BE83/SIP/Info/sipSPIRscmsmAvail: Value returned by check is = 0
009635: Mar 4 12:51:54.271: //32209/742AD833BE83/SIP/Info/sipSPIUaddCcbToUACTable: ****Adding to UAC table.
009636: Mar 4 12:51:54.271: //32209/742AD833BE83/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x4A0D11F4 [email protected]
009637: Mar 4 12:51:54.271: //32209/742AD833BE83/SIP/Info/sipSPIUsetBillingProfile: sipCallId for billing records = [email protected]
009638: Mar 4 12:51:54.271: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 172.24.7.14,Port 5060, Transport 1, SentBy Port 5060
009639: Mar 4 12:51:54.271: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone EST to SIP default timezone = GMT
009640: Mar 4 12:51:54.271: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIgetRegistrarHost: registrar is not configured
009641: Mar 4 12:51:54.271: //32209/742AD833BE83/SIP/Event/sipSPICreateRpid: Received Octet3A=0x80 -> Setting ;screen=no ;privacy=off
SIP: (32209) Group (a= group line) attribute, level 65535 instance 1 not found.
009642: Mar 4 12:51:54.271: //32209/742AD833BE83/SIP/Info/sipSPIGetCallExtensionSupported: anat enabled, src_sdp dont have anat
009643: Mar 4 12:51:54.271: //32209/742AD833BE83/SIP/Info/sipSPISendInvite: Associated container=0x4A43A5F8 to Invite
009644: Mar 4 12:51:54.271: //32209/742AD833BE83/SIP/Transport/sipSPISendInvite: Sending Invite to the transport layer
009645: Mar 4 12:51:54.271: //32209/742AD833BE83/SIP/Transport/sipSPIGetSwitchTransportFlag: Return the Global configuration, Switch Transport is FALSE
009646: Mar 4 12:51:54.271: //32209/742AD833BE83/SIP/Transport/sipSPITransportSendMessage: msg=0x47DBABF4, addr=172.24.7.14, port=5060, sentBy_port=0, is_req=1, transport=1, switch=0, callBack=0x4197823C
009647: Mar 4 12:51:54.271: //32209/742AD833BE83/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
009648: Mar 4 12:51:54.271: //32209/742AD833BE83/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
009649: Mar 4 12:51:54.275: //32209/742AD833BE83/SIP/Transport/sipTransportLogicSendMsg: Set to send the msg=0x47DBABF4
009650: Mar 4 12:51:54.275: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x47DBABF4, addr=172.24.7.14, port=5060, connId=2 for UDP
009651: Mar 4 12:51:54.275: //32209/742AD833BE83/SIP/Info/sentInviteRequest: Sent Invite in state STATE_IDLE
009652: Mar 4 12:51:54.275: //-1/xxxxxxxxxxxx/SIP/Info/sentInviteRequest: Transaction active. Facilities will be queued.
009653: Mar 4 12:51:54.275: //32209/742AD833BE83/SIP/State/sipSPIChangeState: 0x4A0D11F4 : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_SENT_INVITE, SUBSTATE_NONE)
009654: Mar 4 12:51:54.275: //32209/742AD833BE83/SIP/Media/sipSPIProcessRtpSessions: sipSPIProcessRtpSessions
009655: Mar 4 12:51:54.275: //32209/742AD833BE83/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 32209) to the VOIP RTP library
009656: Mar 4 12:51:54.275: //32209/742AD833BE83/SIP/Info/resolve_media_ip_address_to_bind: Media already bound, use existing source_media_ip_addr
009657: Mar 4 12:51:54.275: //32209/742AD833BE83/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 172.25.30.21
009658: Mar 4 12:51:54.275: //32209/742AD833BE83/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
009659: Mar 4 12:51:54.275: //32209/742AD833BE83/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
laddr = 172.25.30.21, lport = 16534, raddr = 0.0.0.0, rport=0, do_rtcp=FALSE
src_callid = 32209, dest_callid = -1, stream type = voice+dtmf, stream direction = RECVONLY
media_ip_addr = - , vrf tableid = 0 media_addr_type = 1
009660: Mar 4 12:51:54.275: //32209/742AD833BE83/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one
009661: Mar 4 12:51:54.275: //32209/742AD833BE83/SIP/Info/sipSPICreateRtpSession: sess: 47DC9F20 do_rtcp:0
009662: Mar 4 12:51:54.275: //32209/742AD833BE83/SIP/Media/sipSPICreateRtpSession: stun is disabled
009663: Mar 4 12:51:54.275: //32209/742AD833BE83/SIP/State/sipSPIChangeStreamState: Stream (callid = 32209) State changed from (STREAM_ADDING) to (STREAM_ACTIVE)
009664: Mar 4 12:51:54.275: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.107.177.95;branch=z9hG4bK+ccab88001db7a774e18a6626099dfafe+10.107.177.95+8
From: "Manager";tag=10.107.177.95+8+1740001+dc7af6ae
To: ""
Date: Thu, 04 Mar 2010 17:51:54 GMT
Call-ID: [email protected]
CSeq: 392882336 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.107.177.95;branch=z9hG4bK+ccab88001db7a774e18a6626099dfafe+10.107.177.95+8
From: "Manager";tag=10.107.177.95+8+1740001+dc7af6ae
To: "";tag=244F8DC4-21CE
Date: Thu, 04 Mar 2010 17:51:54 GMT
Call-ID: [email protected]
CSeq: 392882336 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: ;party=called;screen=no;privacy=off
Contact:
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
009706: Mar 4 12:51:56.479: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [172.24.7.14]:5060
009707: Mar 4 12:51:56.479: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1
009708: Mar 4 12:51:56.479: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000
009709: Mar 4 12:51:56.479: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
009710: Mar 4 12:51:56.479: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Date: Thu, 04 Mar 2010 17:51:54 GMT
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
From: "Manager" ;tag=244F8D1C-1153
Allow-Events: presence, kpml
Supported: replaces
Supported: Geolocation
Content-Length: 211
Require: timer
To: ;tag=65785772-11d5-4cc9-a426-d79b0af58553-56847455
Contact:
Content-Type: application/sdp
Call-ID: [email protected]
Via: SIP/2.0/UDP 172.25.30.21:5060;branch=z9hG4bK298144C
CSeq: 101 INVITE
Session-Expires: 1800;refresher=uas
v=0
o=CiscoSystemsCCM-SIP 2000 1 IN IP4 172.24.7.14
s=SIP Call
c=IN IP4 172.25.30.21
t=0 0
m=audio 19154 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
009711: Mar 4 12:51:56.479: //32209/742AD833BE83/SIP/Info/sipSPICheckResponse: INVITE response with no RSEQ - disable IS_REL1XX
009712: Mar 4 12:51:56.479: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentGTD: No GTD found in inbound container
009713: Mar 4 12:51:56.479: //32209/742AD833BE83/SIP/Info/sipSPIhandle200OKInvite: Transaction active. Facilities will be queued.
009714: Mar 4 12:51:56.479: //32209/742AD833BE83/SIP/Info/sipSPIhandle200OKInvite: *** This ccb is the parent
009715: Mar 4 12:51:56.479: //32208/742AD833BE83/SIP/Info/sipSPIUupdateCcCallIds: Old src/dest ccCallids: -1/-1, new src/dest ccCallids: 32208/32209
009716: Mar 4 12:51:56.479: //32208/742AD833BE83/SIP/Info/sipSPIUupdateCcCallIds: Old streamcallid=-1, new streamcallid=32208
009717: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Error/sipSPIProcessNotifyCallInfoHeader: Call-Info header with for Unsolicited Notify Absent,Disabling Unsolicited Notifies
009718: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/sipSPIUACSessionTimer:
Session-Expires value: 1800 refresher: 2
009719: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_Add_SessionExpiresParamsToContainer: Session-refresh parameters added to container minse = 0 session expire = 1800 refresher = 2
009720: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/sipSPIProcessHistoryInfoHeader: No HI headers recvd from app container
009721: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/sipSPICompareRespMediaInfo: No Comparsion needed as 18x response SDP is either absent or ignored
009722: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/sipSPIDoMediaNegotiation: Number of m-lines = 1
SIP: Attribute mid, level 1 instance 1 not found.
009723: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/resolve_media_ip_address_to_bind: Media already bound, use existing source_media_ip_addr
009724: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 172.25.30.21
009725: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/sipSPIDoAudioNegotiation: Codec (g711ulaw) Negotiation Successful on Static Payload for m-line 1
009726: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/sipSPIDoPtimeNegotiation: One ptime attribute found - value:20
009727: Mar 4 12:51:56.483: //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values :Codec: g711ulaw ptime :20, codecbytes: 160
009728: Mar 4 12:51:56.483: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711ulaw codecbytes :160, ptime: 20
009729: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Media/sipSPIDoPtimeNegotiation: Offered ptime:20, Negotiated ptime:20 Negotiated codec bytes: 160 for codec g711ulaw
009730: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/sipSPIDoDTMFRelayNegotiation: m-line index 1
009731: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(101) could not be reserved.
009732: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/sipSPIDoDTMFRelayNegotiation: RTP-NTE DTMF relay option
009733: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/sipSPIDoDTMFRelayNegotiation: Case of partial named event(NE) match in fmtp list of events.
009734: Mar 4 12:51:56.483: //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: NSE payload from X-cap = 0
009735: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/sip_select_modem_relay_params: X-tmr not present in SDP. Disable modem relay
009736: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/sipSPIGetSDPDirectionAttribute: No direction attribute present or multiple direction attributes that can't be handled for m-line:1 and num-a-lines:0
009737: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/sipSPIDoAudioNegotiation: Codec negotiation successful for media line 1
payload_type=0, codec_bytes=160, codec=g711ulaw, dtmf_relay=rtp-nte
stream_type=voice+dtmf (1), dest_ip_address=172.25.30.21, dest_port=19154
009738: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/State/sipSPIChangeStreamState: Stream (callid = -1) State changed from (STREAM_DEAD) to (STREAM_ADDING)
009739: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Media/sipSPIUpdCallWithSdpInfo:
Preferred Codec : g711ulaw, bytes :160
Preferred DTMF relay : rtp-nte
Preferred NTE payload : 101
Early Media : No
Delayed Media : No
Bridge Done : No
New Media : No
DSP DNLD Reqd : No
009740: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/resolve_media_ip_address_to_bind: Media already bound, use existing source_media_ip_addr
009741: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 172.25.30.21
009742: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_report_media_to_peer:
callId 32209 peer 32208 flags 0x401005 state STATE_RECD_PROCEEDING
009743: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
CallID 32209, sdp 0x47DC33D0 channels 0x4A0D2494
009744: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/copy_channels:
callId 32209 size 468 ptr 0x47E3CD94)
009745: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 0 mline 1
009746: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting codec g711ulaw
009747: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/codec_found:
Codec to be matched: 5
009748: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: ADD AUDIO CODEC 5
009749: Mar 4 12:51:56.483: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711ulaw codecbytes :160, ptime: 20
009750: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Media negotiation done: stream->negotiated_ptime=20,stream->negotiated_codec_bytes=160, coverted ptime=20 stream->mline_index=1, media_ndx=1
009751: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Adding codec 5 ptype 0 time 20, bytes 160 as channel 0 mline 1 ss 1 172.25.30.21:19154
009752: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 101 mline 1
009753: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Call 32209 dtmf ptype 101 nte/oob enabled
009754: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: setting ipip_caps DTMF to RFC2833: callid = 32209, dtmf = 6
009755: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Copy sdp to channel- AFTER CODEC FILTERING: ccb->pld.ipip_caps.codecInfo[channel_ndx].codec = 5
009756: Mar 4 12:51:56.483: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Copy sdp to channel- AFTER CODEC FILTERING: ccb->pld.ipip_caps.codecInfo[channel_ndx].codec = -1
009757: Mar 4 12:51:56.487: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_report_media_to_peer:
callId 32209 flags 0x100 state STATE_RECD_PROCEEDING
009758: Mar 4 12:51:56.487: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_report_media_to_peer:
Report initial call media
009759: Mar 4 12:51:56.487: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_report_media_to_peer: ccb->flags 0xC00018, ccb->pld.flags_ipip 0x403005
009760: Mar 4 12:51:56.487: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_report_media_to_peer: Post CAPS to peer.
009761: Mar 4 12:51:56.487: //32208/742AD833BE83/SIP/Info/ccsip_caps_ind: Entry
009762: Mar 4 12:51:56.487: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_caps_ind:
destCallID=32208, srcCallID=32209,
peer_ccb->call_info.currentLocalName=Manager,
peer_ccb->call_info.currentRemoteName=,
ccb->call_info.currentLocalName=,
ccb->call_info.currentRemoteName=
009763: Mar 4 12:51:56.487: //32208/742AD833BE83/SIP/Info/ccsip_get_rtcp_session_parameters: CURRENT VALUES: stream_callid=32208, current_seq_num=0x9D9
009764: Mar 4 12:51:56.487: //32208/742AD833BE83/SIP/Info/ccsip_get_rtcp_session_parameters: NEW VALUES: stream_callid=32208, current_seq_num=0x0
009765: Mar 4 12:51:56.487: //32208/742AD833BE83/SIP/Info/ccsip_caps_ind: Load DSP with negotiated codec: g711ulaw, Bytes=160
009766: Mar 4 12:51:56.487: //32208/742AD833BE83/SIP/Info/ccsip_caps_ind: Set forking flag to 0x0
009767: Mar 4 12:51:56.487: //32208/742AD833BE83/SIP/Info/sipSPISetDTMFRelayMode: Set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_INBAND_VOICE_AND_OOB
009768: Mar 4 12:51:56.487: //32208/742AD833BE83/SIP/Info/sip_set_modem_caps: Preferred (or the one that came from DSM) modem relay=0, from CLI config=0
009769: Mar 4 12:51:56.487: //32208/742AD833BE83/SIP/Info/sip_set_modem_caps: Disabling Modem Relay...
009770: Mar 4 12:51:56.487: //32208/742AD833BE83/SIP/Info/sip_generate_sdp_xcaps_list: Modem Relay and T38 disabled. X-cap not needed
009771: Mar 4 12:51:56.487: //32208/742AD833BE83/SIP/Info/sip_set_modem_caps: Negotiation already Done. Set negotiated Modem caps and generate SDP Xcap list
009772: Mar 4 12:51:56.487: //32208/742AD833BE83/SIP/Info/sip_set_modem_caps: Modem Relay & Passthru both disabled
009773: Mar 4 12:51:56.487: //32208/742AD833BE83/SIP/Info/sip_set_modem_caps: nse payload = 0, ptru mode = 0, ptru-codec=0, redundancy=0, xid=0, relay=0, sprt-retry=12, latecncy=200, compres-dir=3, dict=1024, strnlen=32
009774: Mar 4 12:51:56.487: //32208/742AD833BE83/SIP/Media/sipSPISetStreamInfo: 0 Active Streams
009775: Mar 4 12:51:56.487: //32208/742AD833BE83/SIP/Media/sipSPISetStreamInfo: Number of active streams is zero (0)!
009776: Mar 4 12:51:56.487: //32208/742AD833BE83/SIP/Media/sipSPISetStreamInfo:
caps.stream_count=0,caps.stream[0].stream_type=0xFFFF, caps.stream_list.xmitFunc=
009777: Mar 4 12:51:56.487: //32208/742AD833BE83/SIP/Media/sipSPISetStreamInfo: ??unknown??, caps.stream_list.context=
009778: Mar 4 12:51:56.487: //32208/742AD833BE83/SIP/Media/sipSPISetStreamInfo: 0x0 (gccb)
009779: Mar 4 12:51:56.487: //32208/742AD833BE83/SIP/Info/ccsip_caps_ind: Load DSP with codec : g711ulaw, Bytes=160, payload = 0
009780: Mar 4 12:51:56.487: //32208/742AD833BE83/SIP/Info/ccsip_caps_ind: ccsip_caps_ind: ccb->pld.flags_ipip = 0x400403
009781: Mar 4 12:51:56.487: //32208/742AD833BE83/SIP/Info/ccsip_caps_ind: No video caps detected in the caps posted by peer leg
009782: Mar 4 12:51:56.487: //32208/742AD833BE83/SIP/Info/ccsip_caps_ind: Calling cc_api_caps_ack()
009783: Mar 4 12:51:56.487: //32209/742AD833BE83/SIP/Info/ccsip_caps_ack: Set forking flag to 0x0
009784: Mar 4 12:51:56.487: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_report_media_to_peer: ccb->flags 0xC00018, ccb->pld.flags_ipip 0x403005
009785: Mar 4 12:51:56.487: //32209/742AD833BE83/SIP/Info/copy_channels:
callId 32209 size 240 ptr 0x47E3CD94)
009786: Mar 4 12:51:56.487: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler:
009787: Mar 4 12:51:56.487: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler: switch(ev.ev_id: 156)
009788: Mar 4 12:51:56.487: //32208/742AD833BE83/SIP/Info/ccsip_event_handler:
ccsip_event_handler: peer ID 32209 chans 0x47E3CD94 event 156 flags 0xC0001C 0x100 0x400403 data 0x47E3CD94
009789: Mar 4 12:51:56.487: //32208/742AD833BE83/SIP/Info/ccsip_event_handler:
ccsip_event_handler: CC_EV_H245_OPEN_CHANNEL_IND: peer ID 32209 chans 0x47E3CD94 event 156 flags 0xC0001C 0x100 0x400403 data 0x47E3CD94
009790: Mar 4 12:51:56.487: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_NEW_MEDIA
009791: Mar 4 12:51:56.487: //32208/742AD833BE83/SIP/Info/ccsip_event_handler:
ccsip_event_handler: set event->type = SIPSPI_EV_CC_NEW_MEDIA!: peer ID 32209 chans 0x47E3CD94 event 156 flags 0xC0001C 0x100 0x400403 data 0x47E3CD94
009792: Mar 4 12:51:56.487: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler: CC_R_SUCCESS_WITH_CONFIRMED
009793: Mar 4 12:51:56.487: //32209/742AD833BE83/SIP/Info/sipSPI_ipip_report_media_to_peer: SIP2SIP, posting channel_ind to peer.
009794: Mar 4 12:51:56.487: //32209/742AD833BE83/SIP/Info/ccsip_update_srtp_caps: 5033: Not Sending NULL SRTP CAPS to SIP LEG
009795: Mar 4 12:51:56.487: //32209/742AD833BE83/SIP/Media/sipSPIUpdCallWithSdpInfo:
Stream type : voice+dtmf
Media line : 1
State : STREAM_ADDING (2)
Stream address type : 1
Callid : 32209
Negotiated Codec : g711ulaw, bytes :160
Nego. Codec payload : 0 (tx), 0 (rx)
Negotiated DTMF relay : rtp-nte
Negotiated NTE payload : 101 (tx), 101 (rx)
Negotiated CN payload : 0
Media Srce Addr/Port : [172.25.30.21]:16534
Media Dest Addr/Port : [172.25.30.21]:19154
009796: Mar 4 12:51:56.487: //32209/742AD833BE83/SIP/Info/sipSPIProcessMediaChanges: sipSPIProcessMediaChanges
009797: Mar 4 12:51:56.487: //32209/742AD833BE83/SIP/Info/sipSPIhandle200OKInvite: ccsip_api_call_connect_media returned: SIP_SUCCESS
009798: Mar 4 12:51:56.487: //32209/742AD833BE83/SIP/State/sipSPIChangeState: 0x4A0D11F4 : State change from (STATE_RECD_PROCEEDING, SUBSTATE_NONE) to (STATE_RECD_PROCEEDING, SUBSTATE_NONE)
009799: Mar 4 12:51:56.487: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQSIG: No QSIG Body found in inbound container
009800: Mar 4 12:51:56.487: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQ931: No RawMsg Body found in inbound container
009801: Mar 4 12:51:56.487: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICreateNewRawMsg: No Data to form The Raw Message
009802: Mar 4 12:51:56.487: //32209/742AD833BE83/SIP/Info/sipSPIShrlCall: Check peer: 11 for Shared-Line call, callid: 32209
009803: Mar 4 12:51:56.487: //32208/742AD833BE83/SIP/Info/sipSPIShrlCall: Check peer: 0 for Shared-Line call, callid: 32208
009804: Mar 4 12:51:56.491: //32209/742AD833BE83/SIP/Info/sipSPIhandle200OKInvite: ccsip_api_call_connected returned: SIP_SUCCESS
009805: Mar 4 12:51:56.491: //32209/742AD833BE83/SIP/Transport/sipSPISendAck: Sending ACK to the transport layer
009806: Mar 4 12:51:56.491: //32209/742AD833BE83/SIP/Transport/sipSPIGetSwitchTransportFlag: Return the Global configuration, Switch Transport is FALSE
009807: Mar 4 12:51:56.491: //32209/742AD833BE83/SIP/Transport/sipSPITransportSendMessage: msg=0x47DBABF4, addr=172.24.7.14, port=5060, sentBy_port=0, is_req=1, transport=1, switch=0, callBack=0x419795EC
009808: Mar 4 12:51:56.491: //32209/742AD833BE83/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
009809: Mar 4 12:51:56.491: //32209/742AD833BE83/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
009810: Mar 4 12:51:56.491: //32209/742AD833BE83/SIP/Transport/sipTransportLogicSendMsg: Set to send the msg=0x47DBABF4
009811: Mar 4 12:51:56.491: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x47DBABF4, addr=172.24.7.14, port=5060, connId=2 for UDP
009812: Mar 4 12:51:56.491: //32209/742AD833BE83/SIP/State/sipSPIChangeState: 0x4A0D11F4 : State change from (STATE_RECD_PROCEEDING, SUBSTATE_NONE) to (STATE_ACTIVE, SUBSTATE_NONE)
009813: Mar 4 12:51:56.491: //32209/742AD833BE83/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4A0D11F4
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 100
Called Number : 4095
Source IP Address (Sig ): 172.25.30.21
Destn SIP Req Addr:Port : 172.24.7.14:5060
Destn SIP Resp Addr:Port : 172.24.7.14:5060
Destination Name : 172.24.7.14
009814: Mar 4 12:51:56.491: //32209/742AD833BE83/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g711ulaw
Negotiated Codec Bytes : 160
Nego. Codec payload : 0 (tx), 0 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 172.25.30.21
Source IP Port (Media): 16534
Destn IP Address (Media): 172.25.30.21
Destn IP Port (Media): 19154
Orig Destn IP Address:Port (Media): [ - ]:0
009815: Mar 4 12:51:56.491: //32209/742AD833BE83/SIP/Info/sipSPICallActive: Transaction Complete. Lock on Facilities released.
009816: Mar 4 12:51:56.491: //32209/742AD833BE83/SIP/Info/sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd
009817: Mar 4 12:51:56.491: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 26
009818: Mar 4 12:51:56.491: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_handle_channel_info:
CCSIP:callID 32208 ft: 1, inc 4, 172.25.30.21:19154, codec 5
009819: Mar 4 12:51:56.491: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_handle_channel_info:
CCSIP:callid 32208 state STATE_SENT_ALERTING
009820: Mar 4 12:51:56.491: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_copy_channelInfo_to_sdp:
callid 32208, channels 0x47E3CD94 caps 0x47E70388
009821: Mar 4 12:51:56.491: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_copy_channelInfo_to_sdp: Peer cap provided: callid = 32208, peer dtmf = 6
009822: Mar 4 12:51:56.491: //32208/742AD833BE83/SIP/Info/codec_found:
Codec to be matched: 5
009823: Mar 4 12:51:56.491: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_copy_channelInfo_to_sdp:
nego mline 1 dtmf 101 ss 1 ret 12
009824: Mar 4 12:51:56.491: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_copy_channelInfo_to_sdp: CCB->pld.flags_ipip 0x400403
009825: Mar 4 12:51:56.491: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_copy_channelInfo_to_sdp: channel_ind/ack payload type 0
009826: Mar 4 12:51:56.491: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711ulaw codecbytes :160, ptime: 20
009827: Mar 4 12:51:56.491: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_handle_channel_info: audio channel_ind
009828: Mar 4 12:51:56.491: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetMediaDirectionForStream: Setting Media direction SENDRECV for stream 1
009829: Mar 4 12:51:56.491: //32208/742AD833BE83/SIP/Info/sipSPIUpdateSrcSdpVariablePart: Setting stream 1 portnum to 18716
009830: Mar 4 12:51:56.491: //32208/742AD833BE83/SIP/Info/sipSPIUpdateSrcSdpVariablePart:
SIP update src sdp, negoitated codec 5, payload type 0
009831: Mar 4 12:51:56.491: //32208/742AD833BE83/SIP/Info/sipSPIProcessMediaChanges: sipSPIProcessMediaChanges
009832: Mar 4 12:51:56.491: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetMediaDirectionForStream: Setting Media direction SENDRECV for stream 1
009833: Mar 4 12:51:56.491: //32208/742AD833BE83/SIP/Info/sipSPIUpdateSrcSdpVariablePart: Setting stream 1 portnum to 18716
009834: Mar 4 12:51:56.491: //32208/742AD833BE83/SIP/Info/sipSPIUpdateSrcSdpVariablePart:
SIP update src sdp, negoitated codec 5, payload type 0
009835: Mar 4 12:51:56.495: //32208/742AD833BE83/SIP/Info/ccsip_bridge: confID = 21265, srcCallID = 32208, dstCallID = 32209
009836: Mar 4 12:51:56.495: //32208/742AD833BE83/SIP/Info/sipSPIUupdateCcCallIds: Old src/dest ccCallids: 32208/32209, new src/dest ccCallids: 32208/32209
009837: Mar 4 12:51:56.495: //32208/742AD833BE83/SIP/Info/sipSPIUupdateCcCallIds: Old streamcallid=32208, new streamcallid=32208
009838: Mar 4 12:51:56.495: //32208/742AD833BE83/SIP/Info/ccsip_gw_set_sipspi_mode: Setting SPI mode to SIP-SIP
009839: Mar 4 12:51:56.495: //32208/742AD833BE83/SIP/Info/ccsip_bridge: xcoder_attached = 0, xmitFunc = 1131939416, ccb xmitFunc = 1131939416
009840: Mar 4 12:51:56.495: //32208/742AD833BE83/SIP/Media/sipSPIProcessRtpSessions: sipSPIProcessRtpSessions
009841: Mar 4 12:51:56.495: //32208/742AD833BE83/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice-only (callid 32208) to the VOIP RTP library
009842: Mar 4 12:51:56.495: //32208/742AD833BE83/SIP/Info/resolve_media_ip_address_to_bind: Media already bound, use existing source_media_ip_addr
009843: Mar 4 12:51:56.495: //32208/742AD833BE83/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 172.25.30.21
009844: Mar 4 12:51:56.495: //32208/742AD833BE83/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
009845: Mar 4 12:51:56.495: //32208/742AD833BE83/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
laddr = 172.25.30.21, lport = 18716, raddr = 10.107.177.95, rport=16746, do_rtcp=TRUE
src_callid = 32208, dest_callid = 32209, stream type = voice-only, stream direction = SENDRECV
media_ip_addr = 10.107.177.95, vrf tableid = 0 media_addr_type = 1
009846: Mar 4 12:51:56.495: //32208/742AD833BE83/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one
009847: Mar 4 12:51:56.495: //32208/742AD833BE83/SIP/Info/sipSPICreateRtpSession: sess: 4A699A58 do_rtcp:1
009848: Mar 4 12:51:56.495: //32208/742AD833BE83/SIP/Media/sipSPICreateRtpSession: stun is disabled
009849: Mar 4 12:51:56.495: //32208/742AD833BE83/SIP/Info/sipSPICreateAndStartRtpTimer:
009850: Mar 4 12:51:56.495: //32208/742AD833BE83/SIP/Info/sipSPICreateAndStartRtpTimer: Media Inactivity Timer is disabled.
009851: Mar 4 12:51:56.495: //32208/742AD833BE83/SIP/Media/sipSPIGetNewLocalMediaDirection:
New Remote Media Direction = SENDRECV
Present Local Media Direction = SENDRECV
New Local Media Direction = SENDRECV
retVal = 0
009852: Mar 4 12:51:56.495: //32208/742AD833BE83/SIP/State/sipSPIChangeStreamState: Stream (callid = 32208) State changed from (STREAM_ADDING) to (STREAM_ACTIVE)
009853: Mar 4 12:51:56.495: //32209/742AD833BE83/SIP/Info/ccsip_bridge: confID = 21265, srcCallID = 32209, dstCallID = 32208
009854: Mar 4 12:51:56.495: //32209/742AD833BE83/SIP/Info/sipSPIUupdateCcCallIds: Old src/dest ccCallids: -1/-1, new src/dest ccCallids: 32209/32208
009855: Mar 4 12:51:56.495: //32209/742AD833BE83/SIP/Info/sipSPIUupdateCcCallIds: Old streamcallid=32209, new streamcallid=32209
009856: Mar 4 12:51:56.495: //32209/742AD833BE83/SIP/Info/ccsip_gw_set_sipspi_mode: Setting SPI mode to SIP-SIP
009857: Mar 4 12:51:56.495: //32209/742AD833BE83/SIP/Info/ccsip_bridge: xcoder_attached = 0, xmitFunc = 1131939416, ccb xmitFunc = 1131939416
009858: Mar 4 12:51:56.495: //32209/742AD833BE83/SIP/Media/sipSPIProcessRtpSessions: sipSPIProcessRtpSessions
009859: Mar 4 12:51:56.495: //32209/742AD833BE83/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 32209) to the VOIP RTP library
009860: Mar 4 12:51:56.495: //32209/742AD833BE83/SIP/Info/resolve_media_ip_address_to_bind: Media already bound, use existing source_media_ip_addr
009861: Mar 4 12:51:56.495: //32209/742AD833BE83/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 172.25.30.21
009862: Mar 4 12:51:56.495: //32209/742AD833BE83/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
009863: Mar 4 12:51:56.495: //32209/742AD833BE83/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
laddr = 172.25.30.21, lport = 16534, raddr = 172.25.30.21, rport=19154, do_rtcp=TRUE
src_callid = 32209, dest_callid = 32208, stream type = voice+dtmf, stream direction = SENDRECV
media_ip_addr = 172.25.30.21, vrf tableid = 0 media_addr_type = 1
009864: Mar 4 12:51:56.495: //32209/742AD833BE83/SIP/Media/sipSPIUpdateRtcpSession: RTP session already created - update
009865: Mar 4 12:51:56.495: //32209/742AD833BE83/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:491045B8
009866: Mar 4 12:51:56.495: //32209/742AD833BE83/SIP/Info/sipSPICreateAndStartRtpTimer:
009867: Mar 4 12:51:56.495: //32209/742AD833BE83/SIP/Info/sipSPICreateAndStartRtpTimer: Media Inactivity Timer is disabled.
009868: Mar 4 12:51:56.495: //32209/742AD833BE83/SIP/Info/sipSPIUpdateRtcpSession:
DTMF inb/oob iwf enabled 101
009869: Mar 4 12:51:56.495: //32209/742AD833BE83/SIP/Media/sipSPIGetNewLocalMediaDirection:
New Remote Media Direction = SENDRECV
Present Local Media Direction = SENDRECV
New Local Media Direction = SENDRECV
retVal = 0
009870: Mar 4 12:51:56.495: //32209/742AD833BE83/SIP/State/sipSPIChangeStreamState: Stream (callid = 32209) State changed from (STREAM_ADDING) to (STREAM_ACTIVE)
009871: Mar 4 12:51:56.495: //32209/742AD833BE83/SIP/Info/ccsip_bridge:
DTMF inb/oob iwf enabled 101
009872: Mar 4 12:51:56.499: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.25.30.21:5060;branch=z9hG4bK2991586
From: "Manager" ;tag=244F8D1C-1153
To: ;tag=65785772-11d5-4cc9-a426-d79b0af58553-56847455
Date: Thu, 04 Mar 2010 17:51:54 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
009873: Mar 4 12:51:56.499: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_call_connect: CCSIP_CALL_CONNECT: ccb ptr 4A0E0A28
009874: Mar 4 12:51:56.499: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_CONNECT
009875: Mar 4 12:51:56.499: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 6
009876: Mar 4 12:51:56.499: //32208/742AD833BE83/SIP/Info/sipSPIAddCiscoGcid: Gcid value not set - not adding header.
009877: Mar 4 12:51:56.499: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_setSessionExpiresParams:
Session refresh values minse = 0 session expire = 1800 refresher = 2
009878: Mar 4 12:51:56.499: //32208/742AD833BE83/SIP/Info/sipSPI_ipip_set_history_info_header: No HI header recvd from container
009879: Mar 4 12:51:56.499: //32208/742AD833BE83/SIP/Info/preprocessConnect: Write sdp_info into msg_body
009880: Mar 4 12:51:56.503: //32208/742AD833BE83/SIP/Info/preprocessConnect: Add msg_body into container 0x4A43A758
009881: Mar 4 12:51:56.503: //32208/742AD833BE83/SIP/Info/sipSPIShrlGetInstanceInfo: Obtained the call instance 0 for non-shared-line '539' with callid: 32208
009882: Mar 4 12:51:56.503: //32208/742AD833BE83/SIP/Event/sipSPICreateRpid: Received Octet3A=0x00 -> Setting ;screen=no ;privacy=off
SIP: (32208) Group (a= group line) attribute, level 65535 instance 1 not found.
009883: Mar 4 12:51:56.503: //32208/742AD833BE83/SIP/Info/sipSPIGetCallExtensionSupported: anat enabled, src_sdp and dest_sdp available, should be a midcall request
009884: Mar 4 12:51:56.503: //32208/742AD833BE83/SIP/Info/sipSPISendInviteResponse: Associated container=0x4A43A758 to Invite Response 200
009885: Mar 4 12:51:56.503: //32208/742AD833BE83/SIP/Transport/sipSPISendInviteResponse: Sending 200OK Response to the Transport Layer
009886: Mar 4 12:51:56.503: //32208/742AD833BE83/SIP/Transport/sipSPITransportSendMessage: msg=0x47DBABF4, addr=10.107.177.95, port=5060, sentBy_port=5060, is_req=0, transport=1, switch=0, callBack=0x41978DB0
009887: Mar 4 12:51:56.503: //32208/742AD833BE83/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
009888: Mar 4 12:51:56.503: //32208/742AD833BE83/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
009889: Mar 4 12:51:56.503: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x47DBABF4, addr=10.107.177.95, port=5060, connId=0 for UDP
009890: Mar 4 12:51:56.503: //32208/742AD833BE83/SIP/Info/sentInviteResponse200: Sent 200Ok for Invite in state STATE_SENT_ALERTING
009891: Mar 4 12:51:56.503: //-1/xxxxxxxxxxxx/SIP/Info/sentInviteResponse200: Transaction active. Facilities will be queued.
009892: Mar 4 12:51:56.503: //32208/742AD833BE83/SIP/State/sipSPIChangeState: 0x4A0E0A28 : State change from (STATE_SENT_ALERTING, SUBSTATE_NONE) to (STATE_SENT_SUCCESS, SUBSTATE_NONE)
009893: Mar 4 12:51:56.503: //- -
JMF error - Format of Stream not supported in RTP Session Manager
java.io.IOException: Format of Stream not supported in RTP Session Manager
at com.sun.media.datasink.rtp.Handler.open(Handler.java:139)
why this erro occors?
I already created the DataSink.
When I try to do this...
dsk.open(); //here the error got
dsk.start(); Code of server of media
I want to sent audio (wav) like a radio, but from file. Without stop to send streaming. PullBufered
*Class Server that you offers Streaming of midia
public class Servidor {
private MediaLocator ml;
private Processor pro;
private javax.media.protocol.DataSource ds;
private DataSink dsk;
private boolean codificado = false;
//start the server service, passing the adress of media
// ex: d:\music\music.wav
// pass the ip and port, to make a server works
public void iniciarServicoServidor(String end,String ip, int porta)
try {
//capture media
capturarMidia(end);
//creates processor
criarProcessor();
// configure the processor
configurarProcessor();
//setContent RAW
descreverConteudoEnviado();
//format the media in right RTP format
formatRTP();
//creat the streaming
criarStreaming();
//configure the server
configurarServidor(ip, porta);
//in this method raise the excepition
iniciarServidor();
//when I try to open the DataSink.open() raises the exception
//java.io.IOException: Format of Stream not supported in RTP Session //Manager
// at com.sun.media.datasink.rtp.Handler.open(Handler.java:139)
} catch (RuntimeException e) {
System.out.println("Houve um erro em iniciarServicoServidor");
e.printStackTrace();
public void capturarMidia(String endereco)
try {
System.out.println("**************************************************************");
System.out.println("Iniciando processo de servidor de multimidia em " + Calendar.getInstance().getTime().toString());
ml = new MediaLocator("file:///" + endereco);
System.out.println("Midia realizada com sucesso.");
System.out.println ("[" + "file:///" + endereco +"]");
} catch (RuntimeException e) {
System.out.println("Houve um erro em capturarMidia");
e.printStackTrace ();
public void criarProcessor()
try {
System.out.println("**************************************************************");
pro = Manager.createProcessor(ml);
System.out.println("Processor criado com sucesso.");
System.out.println("Midia com durcao:" + pro.getDuration().getSeconds());
} catch (NoProcessorException e) {
System.out.println("Houve um erro em criarProcessor");
e.printStackTrace();
} catch (IOException e) {
System.out.println ("Houve um erro em criarProcessor");
e.printStackTrace();
public void configurarProcessor()
try {
System.out.println("**************************************************************");
System.out.println("Processor em estado de configura��o.");
pro.configure();
System.out.println("Processor configurado.");
} catch (RuntimeException e) {
System.out.println("Houve um erro em configurarProcessor");
e.printStackTrace();
public void descreverConteudoEnviado()
try {
System.out.println("**************************************************************");
pro.setContentDescriptor(new ContentDescriptor(ContentDescriptor.RAW));
System.out.println("Descritor de conteudo:" + pro.getContentDescriptor().toString());
} catch (NotConfiguredError e) {
System.out.println("Houve um erro em descreverConteudoEnviado");
e.printStackTrace();
private Format checkForVideoSizes(Format original, Format supported) {
int width, height;
Dimension size = ((VideoFormat)original).getSize();
Format jpegFmt = new Format(VideoFormat.JPEG_RTP);
Format h263Fmt = new Format(VideoFormat.H263_RTP);
if (supported.matches(jpegFmt)) {
// For JPEG, make sure width and height are divisible by 8.
width = (size.width % 8 == 0 ? size.width :
(int)(size.width / 8) * 8);
height = (size.height % 8 == 0 ? size.height :
(int)(size.height / 8) * 8);
} else if (supported.matches(h263Fmt)) {
// For H.263, we only support some specific sizes.
if (size.width < 128) {
width = 128;
height = 96;
} else if ( size.width < 176) {
width = 176;
height = 144;
} else {
width = 352;
height = 288;
} else {
// We don't know this particular format. We'll just
// leave it alone then.
return supported;
return (new VideoFormat(null,
new Dimension(width, height),
Format.NOT_SPECIFIED ,
null,
Format.NOT_SPECIFIED)).intersects(supported);
public void formatRTP()
try {
// Program the tracks.
TrackControl tracks[] = pro.getTrackControls();
Format supported[];
Format chosen;
for (int i = 0; i < tracks.length; i++) {
Format format = tracks.getFormat();
if (tracks[i].isEnabled()) {
supported = tracks[i].getSupportedFormats();
// We've set the output content to the RAW_RTP.
// So all the supported formats should work with RTP.
// We'll just pick the first one.
if (supported.length > 0) {
if (supported[0] instanceof VideoFormat) {
// For video formats, we should double check the
// sizes since not all formats work in all sizes.
chosen = checkForVideoSizes(tracks[i].getFormat(),
supported[0]);
} else
chosen = supported[0];
tracks[i].setFormat(chosen);
System.err.println("Track " + i + " is set to transmit as:");
System.err.println(" " + chosen);
codificado = true;
} else
tracks[i].setEnabled(false);
} else
tracks[i].setEnabled(false);
} catch (RuntimeException e) {
// TODO Auto-generated catch block
e.printStackTrace();
public void tocar()
pro.start();
public void criarStreaming()
try {
System.out.println("**************************************************************");
if (codificado)
System.out.println("Midia codificada...");
System.out.println("Processor entra em estado de realize.");
pro.realize();
System.out.println("Processor realized.");
System.out.println("Adquirindo o streaming a ser enviado.");
ds = pro.getDataOutput();
System.out.println("Streaming adquirido pronto a ser enviado.");
} catch (NotRealizedError e) {
System.out.println("Houve um erro em criarStreaming");
System.out.println(e.getMessage());
e.printStackTrace();
catch (Exception e) {
System.out.println(e.getMessage());
public void configurarServidor(String ip, int porta)
System.out.println("**************************************************************");
String url = "rtp://" + ip + ":" + porta + "/audio/1";
System.out.println("Servidor ira atender em " + url);
MediaLocator mml = new MediaLocator(url);
System.out.println("Localizador de midia ja criado");
try {
System.out.println("Criando um DataSink a ser enviado.");
dsk = Manager.createDataSink(ds, mml);
System.out.println("DataSink criado.");
} catch (NoDataSinkException e) {
e.printStackTrace();
public void iniciarServidor()
try {
System.out.println("**************************************************************");
dsk.open();
System.out.println("Servidor ligado.");
dsk.start();
System.out.println("Servidor iniciado.");
} catch (SecurityException e) {
e.printStackTrace();
} catch (IOException e) {
e.printStackTrace();
Gives that output console.
All methods are executed but the last doesnt works.
The method that open the DataSink.
What can I do?
Iniciando processo de servidor de multimidia em Sun May 13 22:37:02 BRT 2007
Midia realizada com sucesso.
[file:///c:\radio.wav ]
Processor criado com sucesso.
Midia com durcao:9.223372036854776E9
Processor em estado de configura��o.
Processor configurado.
Descritor de conteudo:RAW
Midia codificada...
Processor entra em estado de realize.
Processor realized.
Adquirindo o streaming a ser enviado.
Streaming adquirido pronto a ser enviado.
Servidor ira atender em rtp://127.0.0.1:22000/audio/1
Localizador de midia ja criado
Criando um DataSink a ser enviado.
streams is [Lcom.sun.media.multiplexer.RawBufferMux$RawBufferSourceStream;@a0dcd9 : 1
sink: setOutputLocator rtp://127.0.0.1:22000/audio/1
DataSink criado.
Track 0 is set to transmit as:
unknown, 44100.0 Hz, 16-bit, Stereo, LittleEndian, Signed, 176400.0 frame rate, FrameSize=32 bits
java.io.IOException: Format of Stream not supported in RTP Session Manager
at com.sun.media.datasink.rtp.Handler.open(Handler.java:139)
at br.org.multimidiasi.motor.Servidor.iniciarServidor(Servidor.java:291)
at br.org.multimidiasi.motor.Servidor.iniciarServicoServidor(Servidor.java:43)
at br.org.multimidiasi.motor.ConsoleServidor.main(ConsoleServidor.java:30)
Since already thanks so much.
Exactally in this method raises erros.
Ive tried another formats (avi, mp3) but all with the same error, what I can do?
[code] public void iniciarServidor()
try {
System.out.println("**************************************************************");
dsk.open();
System.out.println("Servidor ligado.");
dsk.start();
System.out.println("Servidor iniciado.");
} catch (SecurityException e) {
e.printStackTrace();
} catch (IOException e) {
e.printStackTrace();
Track 0 is set to transmit as:
unknown, 44100.0 Hz, 16-bit, Stereo, LittleEndian, Signed, 176400.0 frame rate, FrameSize=32 bits
java.io.IOException: Format of Stream not supported in RTP Session Manager
at com.sun.media.datasink.rtp.Handler.open(Handler.java:139)
at br.org.multimidiasi.motor.Servidor.iniciarServidor(Servidor.java:291)
at br.org.multimidiasi.motor.Servidor.iniciarServicoServidor(Servidor.java:43)
at br.org.multimidiasi.motor.ConsoleServidor.main(ConsoleServidor.java:30)unknown, 44100.0 Hz, 16-bit, Stereo,
LittleEndian, Signed, 176400.0 frame rate,
FrameSize=32 bits
java.io.IOException: Format of Stream not supported
in RTP Session Manager
The fact that it doesn't know what the format is
might have to do with the problem. I've had similar
problems, and I've traced it back to missing jars and
codecs. Have you tried running the same code locally
without the transmission to see if you player will
even play the file?Already and it works, I used Player to play it and play normally, I try to make it with the diferents codecs of audio and video, but no sucess. -
Support RTP payload type 8 and 13
I am trying to use JMF to play back recorded RTP streams containing RTP payload type 8 (ALaw) and 13 (CN). It seems these are not supported by default.
Did anybody add support to JMF to support these payload types succesfully? Help is greatly appreciated.Hello
i have a same problem.so i was asking if you solved it or not.
Thanks -
Flash Builder Burrito Supported Android Phones?
Hi there everyone. I am giong to start building an Android phone via Flash Builder Burrito.
The question I wanted to post is, has any tried developing/debugging an Android application on Burrito and connected it with another phone other than the developer phone offered in the Android Developer Marketspace?Any Android 2.2 phone should work, and yes, provided you have the drivers for it installed if you are on Windows. We include the USB drivers for all the common supported phones in your Flash Builder Burrito installation at:
<Adobe Flash Builder Burrito Home>\utilities\drivers\android\
Refer to the documentation that came with Flash Builder Burrito for instructions on installing the drivers.
Also, if you run into any other problems with a particular Android 2.2 device, please let us know!
thanks,
-george -
Help with RTP: Format of Stream not supported in RTP Session Manager
Hello everyone,
I am quite new to JMF and RTP. So far I've succeeded in capturing audio from the microphone and playing it back. However, I failed when I tried to send the stream over using RTP.
Here's my program, all it does is: get a DataSource from the CaptureDevice, create a Processor with that DataSource, convert the tracks in the Processor to one of the RTP formats, and create an RTP SendStream using the Processor's output DataSource.
I can hear sound by creating a Player for the DataSource; however I get errors when I try to create RTP SendStream for the same output DataSource.
Here's my code:
CaptureDeviceInfo cdinfo;
Format fmt = new AudioFormat(AudioFormat.LINEAR, 8000, 8, 1);
Vector deviceList = CaptureDeviceManager.getDeviceList(fmt);
if (deviceList.size() > 0) {
System.out.println("Device Found.");
cdinfo = (CaptureDeviceInfo) deviceList.firstElement();
} else {
System.out.println("No device!");
return;
DataSource ds = Manager.createDataSource(cdinfo.getLocator());
Processor processor = Manager.createProcessor(ds);
StateHelper sh = new StateHelper(processor);
if (!sh.configure(10000)) {
System.out.println("Could not configure...");
System.exit(-1);
// Get the track control objects
TrackControl track[] = processor.getTrackControls();
System.out.println("Number of tracks:" + track.length);
boolean encodingPossible = false;
// Go through the tracks and try to program one of them to outout some "RTP format"
for (int i = 0; i < track.length; i++) {
try {
track.setFormat(new AudioFormat(AudioFormat.DVI_RTP));
encodingPossible = true;
} catch (Exception e) {
// cannot convert
track[i].setEnabled(false);
if (!encodingPossible) {
System.out.println("Could not encode..");
sh.close();
return;
processor.setContentDescriptor(new ContentDescriptor(ContentDescriptor.RAW));
if (!sh.realize(10000)) {
System.out.println("Could not realize...");
System.exit(-1);
System.out.println("Realized...");
DataSource outSource = processor.getDataOutput();
System.out.println(outSource.getContentType());
processor.start();
player = Manager.createRealizedPlayer(outSource);
player.start();
SessionAddress addr = new SessionAddress(InetAddress.getByName("224.144.251.104"), 8194, 4);
manager.initialize(addr);
//manager.addFormat(new AudioFormat(AudioFormat.GSM_RTP), 1);
System.out.println("RTP Session started...");
stream = manager.createSendStream(processor.getDataOutput(), 0);
I get an error on the last line, the error is: javax.media.format.UnsupportedFormatException: Format of Stream not supported in RTP Session Manager And again, if I try to encode the tracks into *AudioFormat.GSM_RTP* instead of *DVI_RTP*, I get a different error on the same line:Exception in thread "AWT-EventQueue-0" java.lang.NullPointerExceptionWell I don't understand what's happening, is there something I need to do before I can use RTP?
Hope you guys help :)Hi,
seems that you are encoding a track to RTP format but outputting a RAW format.
Your encoding section is also a little bit lazy as you don't check supported formats...
Try this between configured and realized state:
// Get the tracks from the processor
TrackControl [] tracks = processor.getTrackControls();
// Do we have at least one track?
if (tracks == null || tracks.length < 1)
return "Couldn't find tracks in processor";
// Set the output content descriptor to RAW_RTP
// This will limit the supported formats reported from
// Track.getSupportedFormats to only valid RTP formats.
ContentDescriptor cd = new ContentDescriptor(ContentDescriptor.RAW_RTP);
processor.setContentDescriptor(cd);
Format supported[];
Format chosen;
boolean atLeastOneTrack = false;
// Program the tracks.
for (int i = 0; i < tracks.length; i++) {
Format format = tracks.getFormat();
log.info("Input format for RTP conversion: " + format);
if (tracks[i].isEnabled()) {
supported = tracks[i].getSupportedFormats();
// We've set the output content to the RAW_RTP.
// So all the supported formats should work with RTP.
if (supported.length > 0) {
if (supported[i] instanceof VideoFormat) {
tracks[i].setEnabled(false);
continue;
else if (supported[i] instanceof AudioFormat) {
// set audio format for RTP transmission
chosen = new AudioFormat(AudioFormat.DVI_RTP);
tracks[i].setFormat(chosen);
tracks[i].setEnabled(true);
atLeastOneTrack = true;
else
tracks[i].setEnabled(false);
else
tracks[i].setEnabled(false);
else
tracks[i].setEnabled(false);
if (!atLeastOneTrack)
return "Couldn't set any of the tracks to a valid RTP format";
The important thing should be theContentDescriptor cd = new ContentDescriptor(ContentDescriptor.RAW_RTP);part.
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