FreeSWITCH

ptime issue when g723 in INVITE sdp

Details

  • Type: Bug Bug
  • Status: Closed Closed
  • Priority: Minor Minor
  • Resolution: Fixed
  • Affects Version/s: None
  • Fix Version/s: 1.2.0
  • Component/s: None
  • Labels:
    None
  • Platform:
    Linux x86_64/gcc
  • CPU Info:
    AMD Athlon(tm) 64 X2 Dual Core Processor 4000+
  • FreeSWITCH GIT Revision:
    HEAD (2012-04-19)
  • Reproduced with GIT HEAD?:
    YES

Description

A provider sending me an SDP with G723 and without ptime makes FS set all codecs defined below G723 in rtpmap to ptime=30, which results in speedy sound when calling applications like playback or voicemail. Bridged calls are fine.

Provider SDP:
v=0
o=CiscoSystemsSIP-GW-UserAgent 2671 9010 IN IP4 193.22.119.34
s=SIP Call
c=IN IP4 193.22.119.13
t=0 0
m=audio 60476 RTP/AVP 18 4 8 0 100 101
c=IN IP4 193.22.119.13
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194,200-202
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 192-194,200-202
a=X-cap: 2 image udptl t38
a=nortpproxy:yes

FreeSWITCH log:
2012-04-19 13:01:34.000552 [DEBUG] sofia_glue.c:4918 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000]
2012-04-19 13:01:34.000552 [DEBUG] sofia_glue.c:4918 Audio Codec Compare [G729:18:8000:20:8000]/[PCMU:0:8000:20:64000]
2012-04-19 13:01:34.000552 [DEBUG] sofia_glue.c:4918 Audio Codec Compare [G723:4:8000:30:6300]/[PCMA:8:8000:20:64000]
2012-04-19 13:01:34.000552 [DEBUG] sofia_glue.c:4918 Audio Codec Compare [G723:4:8000:30:6300]/[PCMU:0:8000:20:64000]
2012-04-19 13:01:34.000552 [DEBUG] sofia_glue.c:4918 Audio Codec Compare [PCMA:8:8000:30:64000]/[PCMA:8:8000:20:64000]
2012-04-19 13:01:34.000552 [DEBUG] sofia_glue.c:4918 Audio Codec Compare [PCMA:8:8000:30:64000]/[PCMU:0:8000:20:64000]
2012-04-19 13:01:34.000552 [DEBUG] sofia_glue.c:4971 Substituting codec PCMA@30i@8000h
2012-04-19 13:01:34.000552 [DEBUG] sofia_glue.c:3017 Set Codec sofia/10.100.1.10/918664023@193.22.119.34 PCMA/8000 30 ms 240 samples 64000 bits

Activity

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François added a comment -
Just in case, FreeSWITCH SDP reply is:
v=0
o=FreeSWITCH 1334823170 1334823171 IN IP4 10.100.1.10
s=FreeSWITCH
c=IN IP4 10.100.1.10
t=0 0
m=audio 10124 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:30
a=sendrecv
Show
François added a comment - Just in case, FreeSWITCH SDP reply is: v=0 o=FreeSWITCH 1334823170 1334823171 IN IP4 10.100.1.10 s=FreeSWITCH c=IN IP4 10.100.1.10 t=0 0 m=audio 10124 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:30 a=sendrecv
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François added a comment -
Apparently, FS detects that it receives 20ms packets so it changes back, which should resolve the issue. The voicemail greeting is still very speedy compared to playing back the recorded wav file.

2012-04-19 13:01:34.130553 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16@8000hz 1 channels 30ms
2012-04-19 13:01:34.280552 [DEBUG] switch_rtp.c:3243 Correct ip/port confirmed.
2012-04-19 13:01:34.550553 [WARNING] mod_sofia.c:1158 Asynchronous PTIME not supported, changing our end from 30 to 20
2012-04-19 13:01:34.550553 [DEBUG] sofia_glue.c:2935 Changing Codec from PCMA@30ms@8000hz to PCMA@20ms@8000hz
2012-04-19 13:01:34.610553 [DEBUG] switch_rtp.c:1555 RE-Starting timer [soft] 160 bytes per 20ms
2012-04-19 13:01:34.610553 [DEBUG] sofia_glue.c:3017 Set Codec sofia/10.100.1.10/918664023@193.22.119.34 PCMA/8000 20 ms 160 samples 64000 bits
2012-04-19 13:01:34.610553 [DEBUG] switch_core_io.c:1095 Engaging Write Buffer at 320 bytes to accommodate 480->320
Show
François added a comment - Apparently, FS detects that it receives 20ms packets so it changes back, which should resolve the issue. The voicemail greeting is still very speedy compared to playing back the recorded wav file. 2012-04-19 13:01:34.130553 [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16@8000hz 1 channels 30ms 2012-04-19 13:01:34.280552 [DEBUG] switch_rtp.c:3243 Correct ip/port confirmed. 2012-04-19 13:01:34.550553 [WARNING] mod_sofia.c:1158 Asynchronous PTIME not supported, changing our end from 30 to 20 2012-04-19 13:01:34.550553 [DEBUG] sofia_glue.c:2935 Changing Codec from PCMA@30ms @8000hz to PCMA@20ms @8000hz 2012-04-19 13:01:34.610553 [DEBUG] switch_rtp.c:1555 RE-Starting timer [soft] 160 bytes per 20ms 2012-04-19 13:01:34.610553 [DEBUG] sofia_glue.c:3017 Set Codec sofia/10.100.1.10/918664023@193.22.119.34 PCMA/8000 20 ms 160 samples 64000 bits 2012-04-19 13:01:34.610553 [DEBUG] switch_core_io.c:1095 Engaging Write Buffer at 320 bytes to accommodate 480->320
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Git added a comment -
Repository: freeswitch
Branch: refs/heads/master
Commit: 6130f61 http://fisheye.freeswitch.org/changelog/freeswitch.git/?cs=6130f61
Updated By: anthm@freeswitch.org

Comment:
FS-4132 try this


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Contact us:
Email: consulting@freeswitch.org
Web: http://www.freeswitch.org
Phone: +1-918-420-9266
Tollfree: +1-877-742-2583
Fax: +1-918-420-9267
iNum: +883 5100 1420 9266


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Git added a comment - Repository: freeswitch Branch: refs/heads/master Commit: 6130f61 http://fisheye.freeswitch.org/changelog/freeswitch.git/?cs=6130f61 Updated By: anthm@freeswitch.org Comment: FS-4132 try this FreeSWITCH Support Contracts and Consulting Services available! Contact us: Email: consulting@freeswitch.org Web: http://www.freeswitch.org Phone: +1-918-420-9266 Tollfree: +1-877-742-2583 Fax: +1-918-420-9267 iNum: +883 5100 1420 9266 Come To ClueCon in August to learn more about FreeSWITCH and Internet Telephony! http://www.cluecon.com
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François added a comment -
It works, thanks!

For info, for the same incoming SDP log now shows:
2012-04-25 12:10:43.000552 [DEBUG] sofia_glue.c:4917 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000]
2012-04-25 12:10:43.000552 [DEBUG] sofia_glue.c:4917 Audio Codec Compare [G729:18:8000:20:8000]/[PCMU:0:8000:20:64000]
2012-04-25 12:10:43.000552 [DEBUG] sofia_glue.c:4917 Audio Codec Compare [G723:4:8000:30:6300]/[PCMA:8:8000:20:64000]
2012-04-25 12:10:43.000552 [DEBUG] sofia_glue.c:4917 Audio Codec Compare [G723:4:8000:30:6300]/[PCMU:0:8000:20:64000]
2012-04-25 12:10:43.000552 [DEBUG] sofia_glue.c:4917 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]

And FS sends SDP:
v=0
o=FreeSWITCH 1335338579 1335338580 IN IP4 10.100.1.10
s=FreeSWITCH
c=IN IP4 10.100.1.10
t=0 0
m=audio 10064 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Show
François added a comment - It works, thanks! For info, for the same incoming SDP log now shows: 2012-04-25 12:10:43.000552 [DEBUG] sofia_glue.c:4917 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] 2012-04-25 12:10:43.000552 [DEBUG] sofia_glue.c:4917 Audio Codec Compare [G729:18:8000:20:8000]/[PCMU:0:8000:20:64000] 2012-04-25 12:10:43.000552 [DEBUG] sofia_glue.c:4917 Audio Codec Compare [G723:4:8000:30:6300]/[PCMA:8:8000:20:64000] 2012-04-25 12:10:43.000552 [DEBUG] sofia_glue.c:4917 Audio Codec Compare [G723:4:8000:30:6300]/[PCMU:0:8000:20:64000] 2012-04-25 12:10:43.000552 [DEBUG] sofia_glue.c:4917 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] And FS sends SDP: v=0 o=FreeSWITCH 1335338579 1335338580 IN IP4 10.100.1.10 s=FreeSWITCH c=IN IP4 10.100.1.10 t=0 0 m=audio 10064 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv
Peter Olsson made changes -
Field Original Value New Value
Status Open [ 1 ] Resolved [ 5 ]
Fix Version/s 1.2.0 [ 10002 ]
Resolution Fixed [ 1 ]
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Due to a long period of inactivity (13 or more days), this issue is due to be automatically close within 24 hours.
If this issue is not actually resolved, please reopen it and add appropriate comments to help developers fix the issue.

Thanks,
  Jira Admin
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Auto Admin added a comment - Due to a long period of inactivity (13 or more days), this issue is due to be automatically close within 24 hours. If this issue is not actually resolved, please reopen it and add appropriate comments to help developers fix the issue. Thanks,   Jira Admin
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Auto Admin added a comment -
Since there has been no change to the status of this issue, it is being closed for inactivity

Thanks,
  Jira Admin


FreeSWITCH Support Contracts and Consulting Services available!

Contact us:
Email: consulting@freeswitch.org
Web: http://www.freeswitch.org
Phone: +1-918-420-9266
Tollfree: +1-877-742-2583
Fax: +1-918-420-9267
iNum: +883 5100 1420 9266


Come To ClueCon in August to learn more about FreeSWITCH and Internet Telephony!
http://www.cluecon.com

Show
Auto Admin added a comment - Since there has been no change to the status of this issue, it is being closed for inactivity Thanks,   Jira Admin FreeSWITCH Support Contracts and Consulting Services available! Contact us: Email: consulting@freeswitch.org Web: http://www.freeswitch.org Phone: +1-918-420-9266 Tollfree: +1-877-742-2583 Fax: +1-918-420-9267 iNum: +883 5100 1420 9266 Come To ClueCon in August to learn more about FreeSWITCH and Internet Telephony! http://www.cluecon.com
Auto Admin made changes -
Status Resolved [ 5 ] Closed [ 6 ]

People

  • Assignee:
    Anthony Minessale II
    Reporter:
    François
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Dates

  • Created:
    Updated:
    Resolved: