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<TITLE>Re: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available!</TITLE>
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<FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>Atleast 2 RTP ports... Possibly more... Also keep in mind that each RTP port requires a handle... So if you have things set to tight you’ll run into issues... Check the ulimits <BR>
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On 7/23/12 7:17 AM, "Brian Foster" <<a href="bdfoster@endigotech.com">bdfoster@endigotech.com</a>> wrote:<BR>
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</SPAN></FONT><BLOCKQUOTE><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>Remember each call is 2 rtp ports.<BR>
<BR>
Brian Foster<BR>
Endigo Computer LLC<BR>
<BR>
Sent from a mobile device.<BR>
<BR>
On Jul 23, 2012 2:28 AM, "Chris Mylonas" <<a href="chris@opencsta.org">chris@opencsta.org</a>> wrote:<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>Sorry to but in,<BR>
<BR>
How many RTP ports have you got configured?<BR>
<BR>
If you only have a range of 5 and you have 5 concurrent calls, then no more RTP ports will be available.<BR>
Doing a `netstat -anp | grep -i udp` should show what ports are open (on linux)<BR>
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On 23/07/2012, at 4:12 PM, curriegrad2004 wrote:<BR>
<BR>
> Tony,<BR>
><BR>
> What can cause this error to happen really? Does FS allow re-use of<BR>
> RTP ports at all given if the media isn't going to the same<BR>
> destination, or doesn't the SDP header define some kind of session<BR>
> tracking mechnaism so ports can be reused?<BR>
><BR>
> On Sun, Jul 22, 2012 at 11:58 AM, Anthony Minessale<BR>
> <<a href="anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>> wrote:<BR>
>> Jira Jira jira<BR>
>><BR>
>> On Jul 22, 2012 12:41 PM, "Phil Quesinberry" <<a href="philq@qsystemsengineering.com">philq@qsystemsengineering.com</a>><BR>
>> wrote:<BR>
>>><BR>
>>> Recently through several of the latest git pulls, FS stops routing calls<BR>
>>> with the following error, usually within a few minutes of making a call<BR>
>>> through Google Voice, not sure if receiving a call through GV will cause the<BR>
>>> problem yet:<BR>
>>><BR>
>>> sofia_glue.c:1083 No RTP ports available!<BR>
>>><BR>
>>> Restarting FS fixes the problem.<BR>
>>><BR>
>>> Before I file the requisite Jira, is anyone aware of this problem or a<BR>
>>> potential configuration issue that might not be compatible with the newer<BR>
>>> builds of FS? This wasn’t happening until the last week or two.<BR>
>>><BR>
>>> Current version we’re running is: FreeSWITCH Version<BR>
>>> 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z<BR>
>>><BR>
>>> Here’s a quick snip from the console when attempting to dial out to show<BR>
>>> the context of the error:<BR>
>>><BR>
>>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash<BR>
>>><BR>
>>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding whether to<BR>
>>> pass zrtp-hash between legs<BR>
>>><BR>
>>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ<BR>
>>> not set, so not propagating zrtp-hash<BR>
>>><BR>
>>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec Compare<BR>
>>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]<BR>
>>><BR>
>>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec<BR>
>>> <a href="sofia/internal/102@192.168.1.6:5060">sofia/internal/102@192.168.1.6:5060</a> <<a href="http://102@192.168.1.6:5060">http://102@192.168.1.6:5060</a>> PCMU/8000 20 ms 160 samples 64000 bits<BR>
>>><BR>
>>> 2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111<BR>
>>> <a href="sofia/internal/102@192.168.1.6:5060">sofia/internal/102@192.168.1.6:5060</a> <<a href="http://102@192.168.1.6:5060">http://102@192.168.1.6:5060</a>> Original read codec set to PCMU:0<BR>
>>><BR>
>>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP.<BR>
>>> Disable 2833 dtmf and switch to INFO<BR>
>>><BR>
>>> 2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports<BR>
>>> available!<BR>
>>><BR>
>>> 2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send signal<BR>
>>> <a href="sofia/internal/102@192.168.1.6:5060">sofia/internal/102@192.168.1.6:5060</a> <<a href="http://102@192.168.1.6:5060">http://102@192.168.1.6:5060</a>> [BREAK]<BR>
>>><BR>
>>> 2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903<BR>
>>> (<a href="sofia/internal/102@192.168.1.6:5060">sofia/internal/102@192.168.1.6:5060</a> <<a href="http://102@192.168.1.6:5060">http://102@192.168.1.6:5060</a>> ) Callstate Change RINGING -> HANGUP<BR>
>>><BR>
>>> 2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup<BR>
>>> <a href="sofia/internal/102@192.168.1.6:5060">sofia/internal/102@192.168.1.6:5060</a> <<a href="http://102@192.168.1.6:5060">http://102@192.168.1.6:5060</a>> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]<BR>
>>><BR>
>>> Regards,<BR>
>>><BR>
>>> Phil Quesinberry<BR>
>>><BR>
>>> Q Systems Engineering, Inc.<BR>
>>><BR>
>>> Electronic Controls and Embedded Systems Development<BR>
>>><BR>
>>> (410) 969-8002 <tel:%28410%29%20969-8002> <BR>
>>><BR>
>>> <a href="http://www.qsystemsengineering.com">http://www.qsystemsengineering.com</a><BR>
>>><BR>
>>><BR>
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>>><BR>
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>><BR>
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