Hello I got problems with choppy voice. I just happens1 time of 10 or something like that. Incorrect call: http://pastebin.freeswitch.org/6479 working call: http://pastebin.freeswitch.org/6481 Any idea what is wrong? Thanks, Jonas
It seems to be related to 20ms vs 30ms ptime. What are the 2 devices and what rev of FS are you on? There was more code added in the last few weeks to smooth out this occurrence. you can also opt to declare your codec prefs at 30ms PCMU@30i instead of just PCMU in the conf. On Thu, Dec 18, 2008 at 11:05 AM, Jonas Gauffin <jonas.gauffin@gmail.com>wrote:
Hello
I got problems with choppy voice. I just happens1 time of 10 or something like that. Incorrect call: http://pastebin.freeswitch.org/6479 working call: http://pastebin.freeswitch.org/6481
Any idea what is wrong?
Thanks, Jonas
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Hello I've checked out the latest trunk, the problem is still left. Im using a Linksys SPA8000 analogue telephone adapter as one device. The other call comes through the sip gateway (from PSTN). I'll try to specify 30ms. Regards, Jonas On Thu, Dec 18, 2008 at 6:20 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
It seems to be related to 20ms vs 30ms ptime.
What are the 2 devices and what rev of FS are you on?
There was more code added in the last few weeks to smooth out this occurrence.
you can also opt to declare your codec prefs at 30ms
PCMU@30i instead of just PCMU in the conf.
On Thu, Dec 18, 2008 at 11:05 AM, Jonas Gauffin <jonas.gauffin@gmail.com>wrote:
Hello
I got problems with choppy voice. I just happens1 time of 10 or something like that. Incorrect call: http://pastebin.freeswitch.org/6479 working call: http://pastebin.freeswitch.org/6481
Any idea what is wrong?
Thanks, Jonas
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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On Thu, Dec 18, 2008 at 11:20:57AM -0600, Anthony Minessale wrote:
It seems to be related to 20ms vs 30ms ptime.
What are the 2 devices and what rev of FS are you on?
There was more code added in the last few weeks to smooth out this occurrence.
This might not be the same issue, but as an additional data point, I can reliably generate audio drop-outs under revision 10725 as follows - it sounds somewhat like a jittery network connection, with occasional glitches. Network problems are not involved, though, as the phone and the machine running FreeSWITCH are both on my desk, connected via my ADSL router. On a Snom 320 SIP phone, select G.722 as the first codec. Call extension 3001 in the default context of the supplied FreeSWITCH configuration, with nobody else calling into the conference, and listen to the resulting audio. Changing the packet size from 20ms to 30ms in the phone's configuration and repeating the test gave me the same result. When I recompile FreeSWITCH and re-install, I'll test this again and post a follow-up if the problem persists.
Riddle me this... what firmware are you running? /b On Dec 19, 2008, at 5:13 PM, Jason White wrote:
On a Snom 320 SIP phone, select G.722 as the first codec.
Call extension 3001 in the default context of the supplied FreeSWITCH configuration, with nobody else calling into the conference, and listen to the resulting audio.
On Fri, Dec 19, 2008 at 05:41:02PM -0600, Brian West wrote:
Please update to 7.1.33 or higher, also I need a pcap of your situation email me a link where I can wget it if off list... I need the rtp and sip traffic. You can do it from the phone or from FreeSWITCH.
I've just upgraded FreeSWITCH to 10889, and I'm definitely having audio issues, but so far only reproduced via remote connections. I'll experiment with the jitter buffer in case it's network related, and I'll run more local tests as well. Yesterday I was experiencing the problem locally, as reported. If it persists and if it appears not to be jitter in the network traffic, I'll run tcpdump and make the output available. Thanks again for the advice.
Do the pcap and show me... because I do not have this issue and I have the exact same phone. /b On Dec 20, 2008, at 1:42 AM, Jason White wrote:
On Fri, Dec 19, 2008 at 05:41:02PM -0600, Brian West wrote:
Please update to 7.1.33 or higher, also I need a pcap of your situation email me a link where I can wget it if off list... I need the rtp and sip traffic. You can do it from the phone or from FreeSWITCH.
I've just upgraded FreeSWITCH to 10889, and I'm definitely having audio issues, but so far only reproduced via remote connections. I'll experiment with the jitter buffer in case it's network related, and I'll run more local tests as well.
Yesterday I was experiencing the problem locally, as reported.
If it persists and if it appears not to be jitter in the network traffic, I'll run tcpdump and make the output available.
Thanks again for the advice.
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is it only the conference? what if you call an extension playing a file instead. On Sat, Dec 20, 2008 at 2:34 PM, Brian West <brian@freeswitch.org> wrote:
Do the pcap and show me... because I do not have this issue and I have the exact same phone.
/b
On Dec 20, 2008, at 1:42 AM, Jason White wrote:
On Fri, Dec 19, 2008 at 05:41:02PM -0600, Brian West wrote:
Please update to 7.1.33 or higher, also I need a pcap of your situation email me a link where I can wget it if off list... I need the rtp and sip traffic. You can do it from the phone or from FreeSWITCH.
I've just upgraded FreeSWITCH to 10889, and I'm definitely having audio issues, but so far only reproduced via remote connections. I'll experiment with the jitter buffer in case it's network related, and I'll run more local tests as well.
Yesterday I was experiencing the problem locally, as reported.
If it persists and if it appears not to be jitter in the network traffic, I'll run tcpdump and make the output available.
Thanks again for the advice.
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
I just tested again, and I myself am now having trouble reproducing it. It happened on multiple occasions yesterday evening, though. It is also easier to reproduce with an actual connection to a remote end-point, but that obviously complicates the situation with potential network issues. I haven't heard it under G.711 though. It can't be system load, as the machine has been mostly idle during my testing/experimentation.
participants (4)
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Anthony Minessale
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Brian West
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Jason White
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Jonas Gauffin