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<TITLE>Re: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available!</TITLE>
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Hey Tony,<BR>
<BR>
Thanks for fixing this...<BR>
<BR>
However, Everyone else, Open the freaking Jira, it doesn’t take that long... Or take the 2 minutes to search jira to see if theres an existing ticket for your problem, and if you have more information add it... <BR>
<BR>
Why? It saves Developers like Tony time in tracking down the issue and getting you a fix... As much as I hate to say it, there could come a time where No Jira ticket == Its not really a problem...<BR>
<BR>
You think tony hasn’t spent a crazy amount of time on FreeSWITCH? Lets do a little math... 50 weeks a year (he’s supposed to get 2 weeks of holiday a year right?) and I know he works at least 40 hours a week on FreeSWITCH so that’s 2000 hours a Year, then factor in that he has been working on it since atleast 2006... So that’s 12000 he has spent working on FreeSWITCH. And that’s at the low end of the estimate... Ohloh.net estimates something like $40million dollars worth of developer time has gone into FreeSWITCH ( <a href="http://www.ohloh.net/p/freeswitch/estimated_cost">http://www.ohloh.net/p/freeswitch/estimated_cost</a> ) most of which is the developers doing their part for the love of the project (ie: not getting paid a dime)... So come on people... Just a little help... That’s all we are asking for here....<BR>
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<BR>
<BR>
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On 7/23/12 11:37 AM, "Anthony Minessale" <<a href="anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>> wrote:<BR>
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</SPAN></FONT><BLOCKQUOTE><FONT FACE="Monaco, Courier New"><SPAN STYLE='font-size:11pt'>commit 524468be7ba5ee0887d2e975e4f131171c601a54<BR>
Author: Anthony Minessale <<a href="anthm@freeswitch.org">anthm@freeswitch.org</a>><BR>
Date: Mon Jul 23 11:36:19 2012 -0500<BR>
<BR>
FS-4373 --resolve this is probably the same issue nobody would file a bug about on the mailing list even after I begged them to from last weekend<BR>
<BR>
<BR>
please please stop filing bugs on the mailing list. I will still work on them and try to fix them but you are making me do way more work.<BR>
Unless everyone wants to pay for support I cannot keep this up.<BR>
<BR>
<BR>
On Mon, Jul 23, 2012 at 8:17 AM, Adam Kelloway <<a href="adam.kelloway@newpace.ca">adam.kelloway@newpace.ca</a>> wrote:<BR>
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I have recently experienced this exact behavior. There doesn't need to be any active calls. It does seem related to the fact that an outbound call was previously made through google voice. A restart also resolves the issue for me as well.<BR>
<BR>
If there is already a JIRA ticket for this, can someone post the link? I can try and help by providing more testing details.<BR>
<BR>
<BR>
<BR>
On 23/07/2012 9:17 AM, Brian Foster wrote:<BR>
<BR>
<BR>
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<BR>
Remember each call is 2 rtp ports.<BR>
<BR>
<BR>
Brian Foster<BR>
Endigo Computer LLC<BR>
<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>
<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>
<BR>
<BR>
<BR>
<BR>
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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