<div dir="ltr">Sorry was away from work a few days, but I should still have the setup which these sip traces came from. I will turn on full console logging and create a Jira ticket with the information asap.<div><br></div>
<div>Thank you!</div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Fri, Oct 4, 2013 at 6:49 PM, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Do you have any logs from this?<div>I have not seen this before and I am not sure what would be causing that.</div>
<div>Full console log with sip traces. POST on JIRA</div><div><br></div></div><div class="gmail_extra"><div><div class="h5">
<br><br><div class="gmail_quote">On Thu, Oct 3, 2013 at 6:00 PM, sangdrax8 <span dir="ltr"><<a href="mailto:sangdrax8@gmail.com" target="_blank">sangdrax8@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<p dir="ltr">Yes head does handle this with out hanging up, but it also inserts it now when the client didn't. So I was wondering why it was adding this which breaks comparability with all older versions of FreeSWITCH which don't insert this.</p>
<div><div>
<div class="gmail_quote">On Oct 3, 2013 6:03 PM, "Sam Russell" <<a href="mailto:sam.h.russell@gmail.com" target="_blank">sam.h.russell@gmail.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">I think Anthony put a patch on HEAD for this - the RFC states that this is acceptable (the a= line should be ignored when the m= line has no port), but there's a regex in recent versions of freeswitch that gets upset when it can't match anything in front of the /0</div>
<div class="gmail_extra"><br><br><div class="gmail_quote">On Fri, Oct 4, 2013 at 2:54 AM, sangdrax8 <span dir="ltr"><<a href="mailto:sangdrax8@gmail.com" target="_blank">sangdrax8@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">It seems that the current head has actually started producing this "rtpmap: 96 /0", when my client devices do not include it in the sdp. This causes problems when I am trying to upgrade one switch at a time in my production environment. I assume (will test here shortly) that if all switches are on the newer version, this wouldn't be an issue as the indicated patch should handle it. <div>
<br></div><div>I would really like to be able to upgrade my boxes individually, but if an upgraded box add's this to the SDP, my non-upgraded boxes will kill the call.</div><div><br></div><div>The following is an example SDP trace from my latest head (10.01.2013) showing the SDP from the client device is normal, and then what is sent to the other freeswitch (my currently non upgraded one) includes this problem causing rtpmap. I should note that this seems to only be the case when I activate zrtp on the clients, which causes my dial plan to set proxy_media=true. </div>
<div><br></div><div>I haven't opened a bug in jira, I wanted to confirm if this rtpmap is in fact a bug. If there is a good reason it is there now and isn't in my older freeswitch versions, then I may have to attempt to upgrade all at one time.</div>
<div><br></div><div><div><div>recv 1072 bytes from tls/[1.1.1.1]:50272 at 13:42:16.509540:</div><div> ------------------------------------------------------------------------</div><div> SIP/2.0 200 OK</div><div> Via: SIP/2.0/TLS 2.2.2.2;branch=z9hG4bK5N858D7BX46je</div>
<div> Contact: <sip:<a href="tel:15715550001" value="+15715550001" target="_blank">15715550001</a>@192.168.10.136:60068;transport=tls></div><div> From: "Test 1" <sip:<a href="tel:17035550001" value="+17035550001" target="_blank">17035550001</a>@securevoice>;tag=jSmQZ67etF8pa</div>
<div> Call-ID: 6fe45089-a6d4-1231-2aac-005056ac002a</div>
<div> CSeq: 50067799 INVITE</div><div> To: <sip:<a href="tel:15715550001" value="+15715550001" target="_blank">15715550001</a>@192.168.10.136:60068;rinstance=A5E7AF45;transport=tls>;tag=8508A76BFA383C2109766C45F68BCEFB</div>
<div> Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY, MESSAGE</div>
<div> Supported: replaces, path</div><div> Content-Type: application/sdp</div><div> Content-Length: 534</div><div><br></div><div> v=0</div><div> o=- 98627 19401 IN IP4 192.168.10.136</div><div> s=menduco</div>
<div> c=IN IP4 192.168.10.136</div><div> t=0 0</div><div> m=audio 58620 RTP/SAVP 9 101</div><div> a=rtpmap:101 telephone-event/8000</div><div> a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Df3mAFtpRclAzm98vfNWAoWqi6WaA3hfWs0e7CSi</div>
<div> a=ptime:30</div><div> a=fmtp:101 0-15</div><div> a=zrtp-hash:1.10 99B821F2DBB96DFDC234440977DC3956441DEC6CDAE690B1B06C740F6EBC29B0</div><div> a=sendrecv</div><div> m=audio 0 RTP/AVP 9 0 101</div><div> a=rtpmap:101 TELEPHONE-EVENT/8000</div>
<div> a=ptime:30</div><div> a=fmtp:101 0-15</div><div> a=zrtp-hash:1.10 18489BD8B5526F4C92AB6EB2A04022C25E8CB5D7E58D4E633FC4545783447F8C</div><div> ------------------------------------------------------------------------</div>
<div>send 472 bytes to tls/[1.1.1.1]:50272 at 13:42:16.510594:</div><div> ------------------------------------------------------------------------</div><div> ACK sip:<a href="tel:15715550001" value="+15715550001" target="_blank">15715550001</a>@192.168.10.136:60068;transport=tls SIP/2.0</div>
<div> Via: SIP/2.0/TLS 2.2.2.2;branch=z9hG4bK6y1ya9QFtDX5S</div><div> Max-Forwards: 70</div><div> From: "Test 1" <sip:<a href="tel:17035550001" value="+17035550001" target="_blank">17035550001</a>@securevoice>;tag=jSmQZ67etF8pa</div>
<div> To: <sip:<a href="tel:15715550001" value="+15715550001" target="_blank">15715550001</a>@192.168.10.136:60068;rinstance=A5E7AF45;transport=tls>;tag=8508A76BFA383C2109766C45F68BCEFB</div>
<div> Call-ID: 6fe45089-a6d4-1231-2aac-005056ac002a</div><div> CSeq: 50067799 ACK</div><div> Contact: <sip:mod_sofia@2.2.2.2:5061;transport=tls></div><div> Content-Length: 0</div><div>------------------------------------------------------------------------</div>
<div>send 1516 bytes to tls/[10.110.11.31]:47793 at 13:42:16.516187:</div><div> ------------------------------------------------------------------------</div><div> SIP/2.0 200 OK</div><div> Via: SIP/2.0/TLS 10.110.11.31:5071;branch=z9hG4bKD97QXp3XZgX4a;rport=47793</div>
<div> From: "Test 1" <<a href="mailto:sip%3A17035550001@10.110.11.31" target="_blank">sip:17035550001@10.110.11.31</a>>;tag=DU0S42e40FNpg</div><div> To: <sip:<a href="tel:15715550001" value="+15715550001" target="_blank">15715550001</a>@dcprotopop1.private.sec:5071;transport=tls>;tag=tcXjHQ6gme91S</div>
<div> Call-ID: 6fd5bd89-a6d4-1231-8291-005056ac002c</div><div> CSeq: 50067799 INVITE</div><div> Contact: <sip:<a href="tel:15715550001" value="+15715550001" target="_blank">15715550001</a>@10.110.11.32:5071;transport=tls></div>
<div> User-Agent: FreeSWITCH-mod_sofia/1.5.6b+git~20131001T160636Z~6d2280df08</div>
<div> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE</div><div> Supported: timer, precondition, path, replaces</div><div> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer</div>
<div> Content-Type: application/sdp</div><div> Content-Disposition: session</div><div> Content-Length: 408</div><div> X-FS-Display-Name: Outbound Call</div><div> X-FS-Display-Number: sip:<a href="tel:15715550001" value="+15715550001" target="_blank">15715550001</a>@dcprotopop1.private.sec</div>
<div> X-FS-Support: update_display,send_info</div><div> Remote-Party-ID: "Outbound Call" <sip:<a href="tel:15715550001" value="+15715550001" target="_blank">15715550001</a>@dcprotopop1.private.sec>;party=calling;privacy=off;screen=no</div>
<div><br></div><div> v=0</div>
<div> o=FreeSWITCH 1167543025 1167543026 IN IP4 10.110.11.32</div><div> s=FreeSWITCH</div><div> c=IN IP4 10.110.11.32</div><div> t=0 0</div><div> m=audio 27868 RTP/SAVP 9 101</div><div> a=rtpmap:101 telephone-event/8000</div>
<div> a=fmtp:101 0-15</div><div> a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:Df3mAFtpRclAzm98vfNWAoWqi6WaA3hfWs0e7CSi</div><div> a=ptime:30</div><div> a=zrtp-hash:1.10 99B821F2DBB96DFDC234440977DC3956441DEC6CDAE690B1B06C740F6EBC29B0</div>
<div> m=audio 0 RTP/AVP 9 0 96</div><div> a=rtpmap:96 /0</div></div></div><div><br></div></div>
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