Hello,<br><br>If you are a codec/ptime person, great. I've been "round and round" with the vendor on this and and not making much headway.<br><br>We've an FS switch that has two numbers registered with our provider. That provider is, in turn, getting service from another gateway provider of their own.<br>
<br>On our FS, one DID works with no issues and has for more than a year. They other recently started having ptime mismatch issues making it unusable. Tried Scrooge setting to no avail. <br><br>For the daring (and merciful), here's FS output for a "bad" call and then a "good" one (just a little output from the start):<br>
<br>
<br><div>2011-06-02 14:12:06.756170 [DEBUG] switch_core_state_machine.c:397 (sofia/external/<a href="http://8154835414@208.94.159.10:5060/" target="_blank">8151231234@xxx.xxx.xxx.xxx:5060</a>) Running State Change CS_NEW </div>
<div>
2011-06-02 14:12:06.756170 [DEBUG] sofia.c:3210 Channel sofia/external/<a href="http://8154835414@208.94.159.10:5060/" target="_blank">8151231234@xxx.xxx.xxx.xxx:5060</a> entering state [received][100] </div>
<div>2011-06-02
14:12:06.756170 [DEBUG] sofia.c:3217 Remote SDP:
</div>
<div>v=0
</div><div>o=Acme_UAS 0 1 IN IP4 xxx.xxx.xxx.xxx <br></div>
<div>s=SIP Media Capabilities
</div><div>c=IN IP4 yyy.yyy.yyy.yyy
<br></div>
<div>t=0 0
</div><div>m=audio 52024 RTP/AVP 0 18 101
</div>
<div>a=rtpmap:0 PCMU/8000
</div><div>a=rtpmap:18 G729/8000
</div>
<div>a=rtpmap:101 telephone-event/8000
</div><div><b>a=maxptime:30 </b>
</div>
<div>
</div><div>2011-06-02 14:12:06.756170 [DEBUG] switch_core_state_machine.c:403 (sofia/external/<a href="http://8154835414@208.94.159.10:5060/" target="_blank">8151231234@xxx.xxx.xxx.xxx:5060</a>) State NEW </div>
<div>2011-06-02 14:12:06.756170 [DEBUG] sofia_glue.c:3081 <b>Audio Codec Compare [PCMU:0:8000:30]/[PCMU:0:8000:</b><b>20]</b> </div>2011-06-02
14:12:06.756170 [DEBUG] sofia_glue.c:3092 Bah HUMBUG! Sticking with
PCMU@8000h@20i <br> <br><div>2011-06-02 14:12:06.756170 [DEBUG] sofia_glue.c:2039 Set Codec sofia/external/<a href="http://8154835414@208.94.159.10:5060/" target="_blank">8151231234@xxx.xxx.xxx.xxx:5060</a> PCMU/8000 20 ms 160 samples </div>
2011-06-02 14:12:06.756170 [DEBUG] sofia_glue.c:3041 Set 2833 dtmf payload to 101<br><br><br><br>Note that whilst it says its sticking with ptime of 20 (bolded above),<u> <i>our vendor in-the-middle sees us as still using ptime=30</i></u>. Weird. But, on the same switch with different DID, we get the following "good" result:<br>
<br><br><br><div>2011-06-02 14:15:15.135193 [DEBUG] switch_core_state_machine.c:397 (sofia/external/<a href="http://8159535620@208.94.157.10:5060/" target="_blank">8151231234@xxx.xxx.xxx.xxx:5060</a>) Running State Change CS_NEW </div>
<div>
2011-06-02 14:15:15.135193 [DEBUG] sofia.c:3210 Channel sofia/external/<a href="http://8159535620@208.94.157.10:5060/" target="_blank">8151231234@ xxx.xxx.xxx.xxx:5060</a> entering state [received][100] </div>
<div>2011-06-02
14:15:15.135193 [DEBUG] sofia.c:3217 Remote SDP:
</div>
<div>v=0
</div><div>o=Acme_UAS 0 1 IN IP4 xxx.xxx.xxx.xxx <br></div>
<div>s=SIP Media Capabilities
</div><div>c=IN IP4 yyy.yyy.yyy.yyy
<br></div>
<div>t=0 0
</div><div>m=audio 52052 RTP/AVP 0 18 101
</div>
<div>a=rtpmap:0 PCMU/8000
</div><div>a=rtpmap:18 G729/8000
</div>
<div>a=rtpmap:101 telephone-event/8000
</div><div><b>a=maxptime:20</b>
</div>
<div>
</div><div>2011-06-02 14:15:15.135193 [DEBUG] switch_core_state_machine.c:403 (sofia/external/<a href="http://8159535620@208.94.157.10:5060/" target="_blank">8151231234@xxx.xxx.xxx.xxx.10:5060</a>) State NEW </div>
<div>2011-06-02 14:15:15.135193 [DEBUG] sofia_glue.c:3081 <b>Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:</b><b>20]</b> </div><div>2011-06-02
14:15:15.135193 [DEBUG] sofia_glue.c:3092 Bah HUMBUG! Sticking with
PCMU@8000h@20i
</div>
<div>2011-06-02 14:15:15.135193 [DEBUG] sofia_glue.c:2039 Set Codec sofia/external/<a href="http://8159535620@208.94.157.10:5060/" target="_blank">8151231234@xxx.xxx.xxx.xxx:5060</a> PCMU/8000 20 ms 160 samples </div>
2011-06-02 14:15:15.135193 [DEBUG] sofia_glue.c:3041 Set 2833 dtmf
payload to 101<br><br><br>Naturally, the "far end" vendor says they think its a "FreeSwitch problem". Huh? For my part, I'd just like to FORCE a ptime of 20 in all cases, no matter the codec, for incoming calls, either DID. Tried, but so far, no go. <br>
<br>A packet capture showed "bad" call coming in with ptime max of 30, we respond that 30 is OK but them re-invite with ptime set to 20, but it doesn't seem to "catch". On the good DID, they proffer ptime max of 30, we respond with ptime of 20 and life is good. <br>
<br>Sorry to post again on this. I have put in some hours but still same results so far. Any suggestions much appreciated. <br><br>Regards,<br><br>Mike