[asterisk-bugs] [JIRA] (ASTERISK-24691) Asterisk tries to transcode between g722 & h264
Mark Farmer (JIRA)
noreply at issues.asterisk.org
Thu Jan 22 10:56:34 CST 2015
[ https://issues.asterisk.org/jira/browse/ASTERISK-24691?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Mark Farmer updated ASTERISK-24691:
-----------------------------------
Attachment: myDebugLog.gz
Please find attached debug log file.
One of our engineers has extracted the following from the debug log, hopefully it will help to identify & solve the problem.
{noformat}
[2015-01-21 15:08:10] DEBUG[12273][C-00000031] chan_sip.c: Processing session-level SDP b=AS:128... UNSUPPORTED OR FAILED.
[2015-01-21 15:08:10] DEBUG[12273][C-00000031] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
[2015-01-21 15:08:10] DEBUG[12273][C-00000031] chan_sip.c: Processing session-level SDP a=sendrecv... OK.
[2015-01-21 15:08:10] VERBOSE[12273][C-00000031] chan_sip.c: Found RTP audio format 9
[2015-01-21 15:08:10] DEBUG[12273][C-00000031] rtp_engine.c: Setting payload 9 (0x7fd1680a71e8) based on m type on 0x7fd11bed6840
[2015-01-21 15:08:10] VERBOSE[12273][C-00000031] chan_sip.c: Found RTP audio format 101
[2015-01-21 15:08:10] DEBUG[12273][C-00000031] rtp_engine.c: Setting payload 101 (0x7fd168513a48) based on m type on 0x7fd11bed6840
[2015-01-21 15:08:10] VERBOSE[12273][C-00000031] chan_sip.c: Found audio description format G722 for ID 9
[2015-01-21 15:08:10] DEBUG[12273][C-00000031] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK.
[2015-01-21 15:08:10] VERBOSE[12273][C-00000031] chan_sip.c: Found audio description format telephone-event for ID 101
[2015-01-21 15:08:10] DEBUG[12273][C-00000031] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[2015-01-21 15:08:10] DEBUG[12273][C-00000031] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[2015-01-21 15:08:10] VERBOSE[12273][C-00000031] chan_sip.c: Found RTP video format 99
[2015-01-21 15:08:10] DEBUG[12273][C-00000031] rtp_engine.c: Setting payload 99 (0x7fd168513a48) based on m type on 0x7fd11bed67d0
[2015-01-21 15:08:10] VERBOSE[12273][C-00000031] chan_sip.c: Found video description format H264 for ID 99
[2015-01-21 15:08:10] DEBUG[12273][C-00000031] chan_sip.c: Processing media-level (video) SDP a=rtpmap:99 H264/90000... OK.
[2015-01-21 15:08:10] DEBUG[12273][C-00000031] chan_sip.c: Processing media-level (video) SDP a=fmtp:99 profile-level-id=42800d... OK.
[2015-01-21 15:08:10] DEBUG[12273][C-00000031] chan_sip.c: Processing media-level (video) SDP a=sendrecv... UNSUPPORTED OR FAILED.
[2015-01-21 15:08:10] VERBOSE[12273][C-00000031] chan_sip.c: Capabilities: us - (g722|ulaw|alaw|h264), peer - audio=(g722)/video=(h264)/text=(nothing), combined - (g722|h264)
[2015-01-21 15:08:10] VERBOSE[12273][C-00000031] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2015-01-21 15:08:10] DEBUG[12273][C-00000031] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7fd1b806fc98'
[2015-01-21 15:08:10] VERBOSE[12273][C-00000031] chan_sip.c: Peer audio RTP is at port 192.168.5.49:10028
[2015-01-21 15:08:10] DEBUG[12273][C-00000031] rtp_engine.c: Copying payload 9 (0x7fd168484d28) from 0x7fd11bed6840 to 0x7fd1b806fdd8
[2015-01-21 15:08:10] DEBUG[12273][C-00000031] rtp_engine.c: Copying payload 101 (0x7fd1680a71e8) from 0x7fd11bed6840 to 0x7fd1b806fdd8
[2015-01-21 15:08:10] DEBUG[12273][C-00000031] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7fd1b806fc98'
[2015-01-21 15:08:10] DEBUG[12273][C-00000031] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7fd1b80d8ff8'
[2015-01-21 15:08:10] VERBOSE[12273][C-00000031] chan_sip.c: Peer video RTP is at port 192.168.5.49:10030
[2015-01-21 15:08:10] DEBUG[12273][C-00000031] rtp_engine.c: Copying payload 99 (0x7fd168279fe8) from 0x7fd11bed67d0 to 0x7fd1b80d9138
[2015-01-21 15:08:10] DEBUG[12273][C-00000031] chan_sip.c: We're settling with these formats: (g722|h264)
[2015-01-21 15:08:10] DEBUG[12273][C-00000031] chan_sip.c: We have an owner, now see if we need to change this call
[2015-01-21 15:08:10] DEBUG[12273][C-00000031] chan_sip.c: Setting native formats after processing SDP. peer joint formats (g722|h264), old nativeformats (g722|h264)
[2015-01-21 15:08:10] DEBUG[12273][C-00000031] chan_sip.c: Updating call counter for outgoing call
[2015-01-21 15:08:10] DEBUG[12211] devicestate.c: No provider found, checking channel drivers for SIP - 6046
[2015-01-21 15:08:10] DEBUG[12273][C-00000031] chan_sip.c: Strict routing enforced for session 674234be5bb2d30827882c3e4d53dcdb at 213.216.146.208:5060
[2015-01-21 15:08:10] DEBUG[12211] chan_sip.c: Checking device state for peer 6046
[2015-01-21 15:08:10] DEBUG[12211] devicestate.c: Changing state for SIP/6046 - state 2 (In use)
[2015-01-21 15:08:10] VERBOSE[12273][C-00000031] chan_sip.c: Transmitting (NAT) to 195.59.152.66:20626:
{noformat}
> Asterisk tries to transcode between g722 & h264
> -----------------------------------------------
>
> Key: ASTERISK-24691
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-24691
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_sip/CodecHandling
> Affects Versions: 13.1.0
> Environment: CentOS 6.6
> Reporter: Mark Farmer
> Assignee: Mark Farmer
> Attachments: myDebugLog.gz
>
>
> With the following codecs on SIP peers:
> disallow = all
> allow = g722,ulaw,alaw,h264
> When we try to originate a call via AMI or transfer a call that was answered via a queue, Asterisk seems to try to use h264 and transcode between g722 & h264.
> Console output:
> {noformat}
> -- Called SIP/6059
> -- Local/6059 at GageAgent-00000032;1 connected line has changed. Saving it until answer for SIP/ph-sip03-gn-lon1-0000008b
> -- SIP/6059-0000008c is ringing
> -- Local/6059 at GageAgent-00000032;1 is ringing
> -- SIP/6059-0000008c answered Local/6059 at GageAgent-00000032;2
> -- Local/6059 at GageAgent-00000032;1 answered SIP/ph-sip03-gn-lon1-0000008b
> -- Channel Local/6059 at GageAgent-00000032;2 joined 'simple_bridge' basic-bridge <2765d56b-fcb8-46b5-8d83-7866feea3196>
> -- Stopped music on hold on SIP/ph-sip03-gn-lon1-0000008b
> -- Channel SIP/6059-0000008c joined 'simple_bridge' basic-bridge <2765d56b-fcb8-46b5-8d83-7866feea3196>
> -- Channel SIP/ph-sip03-gn-lon1-0000008b joined 'simple_bridge' basic-bridge <70114587-b1d8-46a0-9f4e-ad77ed10da51>
> -- Channel Local/6059 at GageAgent-00000032;1 joined 'simple_bridge' basic-bridge <70114587-b1d8-46a0-9f4e-ad77ed10da51>
> -- SIP/6032-0000008e is ringing
> -- SIP/6032-0000008e answered SIP/6059-0000008d
> -- Channel SIP/6059-0000008d joined 'simple_bridge' basic-bridge <92d9bf20-c496-4b6c-86bf-2919395a56bf>
> -- Channel SIP/6032-0000008e joined 'simple_bridge' basic-bridge <92d9bf20-c496-4b6c-86bf-2919395a56bf>
> -- Channel Local/6059 at GageAgent-00000032;2 left 'simple_bridge' basic-bridge <2765d56b-fcb8-46b5-8d83-7866feea3196>
> -- Channel Local/6059 at GageAgent-00000032;2 swapped with SIP/6059-0000008d into 'native_rtp' basic-bridge <92d9bf20-c496-4b6c-86bf-2919395a56bf>
> -- Stopped music on hold on Local/6059 at GageAgent-00000032;2
> -- Channel SIP/6059-0000008d left 'native_rtp' basic-bridge <92d9bf20-c496-4b6c-86bf-2919395a56bf>
> -- Channel SIP/6059-0000008c left 'simple_bridge' basic-bridge <2765d56b-fcb8-46b5-8d83-7866feea3196>
> == Spawn extension (DLPN_All, 6032, 50006) exited non-zero on 'SIP/6059-0000008d'
> [2015-01-15 09:29:05] WARNING[19055][C-0000007f]: channel.c:5070 ast_write: Codec mismatch on channel SIP/6032-0000008e setting write format to ulaw from slin16 native formats (g722|h264)
> [2015-01-15 09:29:05] WARNING[19055][C-0000007f]: channel.c:5070 ast_write: Codec mismatch on channel SIP/6032-0000008e setting write format to slin16 from ulaw native formats (g722|h264)
> -- Registered SIP '6147' at xxx.xxx.xxx.xxx:61292
> -- Registered SIP '6147' at xxx.xxx.xxx.xxx:8029
> -- Channel SIP/ph-sip03-gn-lon1-0000008b left 'simple_bridge' basic-bridge <70114587-b1d8-46a0-9f4e-ad77ed10da51>
> -- Channel Local/6059 at GageAgent-00000032;1 left 'simple_bridge' basic-bridge <70114587-b1d8-46a0-9f4e-ad77ed10da51>
> == Spawn extension (voicemenu-custom-19, s, 5) exited non-zero on 'SIP/ph-sip03-gn-lon1-0000008b'
> -- Channel Local/6059 at GageAgent-00000032;2 left 'simple_bridge' basic-bridge <92d9bf20-c496-4b6c-86bf-2919395a56bf>
> == MixMonitor close filestream (mixed)
> == End MixMonitor Recording SIP/ph-sip03-gn-lon1-0000008b
> {noformat}
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