<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><div>Hi,</div><div><br></div><div>I'm having a strange behavior with the FS when I'm using it with "inboud-late-negotiation=true" and with the both scenarios "proxy-media=true" or "bypass-media=true". The FS is acting as a pseudo proxy (I know that it is not intend for that).</div><div><br></div><div>The configuration is similar to this: </div><div><br></div><div>[Endpoints] <= a => [FS1] <= b => [FS2]</div><div><br></div><div>Where FS1 is acting as a proxy and registrar. The other one will simply handle the calls between endpoints or with a Gateway.</div><div><br></div><div>Basically the problem is when I'm trying to call to an AddPac Endpoint, then FS2 sends a call to the FS1 and this one do the proxy or bypass process. Everything works fine with most of the UA (Grandstream/Sipura/Linksys/Xlite/Zoiper/etc) but with the AddPac the FS1 is returning a Payload 96 which seams to be wrong. The only different (In Comparison to the other UA) that I've seen is the "a=ptime:20" in the 200/OK/SDP witch is teorically right. I've seen some posts with a similar issue las year: <a href="http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg01468.html">http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg01468.html</a></div><div><br></div><div>Another interesting point is when the media is handle by FS1 ("proxy-media=false"|"bypass-media=false") everything works fine. Also, what is more weird is when the call is generated by the AddPac the SDP everything works fine too.</div><div><br></div><div>Probably is an AddPac issue, due is the only one failing, however since I have not found something wrong in the SIP capture I'm starting to have my doubts...</div><div><br></div><div>I'm running Freeswitch 1.0.4pre9.</div><div><br></div><div>Here is a small flow of the call:</div><div><br></div><div>1) FS2 INVITE TO FS1</div><div>2) FS1 INVITE TO AddPac Endpoint</div><div>3) AddPac Endpoint responds 200 with apparently correct SDP.</div><div>4) FS1 responds 200/OK with an Incorrect Payload</div><div><br></div><div>Here are the corresponding SIP packet trace with the previous call flow:</div><div><br></div><div>Packet 1</div><div>=======</div><div><div>INVITE <a href="sip:9499@">sip:9499@</a>[FS1 IP] SIP/2.0</div><div>Via: SIP/2.0/UDP [FS2 IP]:5060;branch=z9hG4bK60c74dbf;rport</div><div>From: "[Source Number]" <sip:[Source Number]@[FS2 IP]>;tag=as3bd06925</div><div>To: <<a href="sip:9499@">sip:9499@</a>[FS1 IP]></div><div>Contact: <sip:[Source Number]@[FS2 IP]></div><div>Call-ID: 14b07c2014d3a0aa7719efa03d979eed@[FS2 IP]</div><div>CSeq: 102 INVITE</div><div>User-Agent: Legacy</div><div>Max-Forwards: 70</div><div>Remote-Party-ID: "[Source Number]" <sip:[Source Number]@[FS2 IP]>;privacy=off;screen=no</div><div>Date: Mon, 28 Sep 2009 15:28:19 GMT</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</div><div>Supported: replaces</div><div>Content-Type: application/sdp</div><div>Content-Length: 240</div><div><br></div><div>v=0</div><div>o=root 16330 16330 IN IP4 [FS2 IP]</div><div>s=session</div><div>c=IN IP4 [FS2 IP]</div><div>t=0 0</div><div>m=audio 12558 RTP/AVP 0 101</div><div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div><div>a=silenceSupp:off - - - -</div><div>a=ptime:20</div><div>a=sendrecv</div><div><br></div><div><br></div></div><div>Packet 2</div><div>=======</div><div><div>INVITE <a href="sip:9499@">sip:9499@</a>[AddPac IP] SIP/2.0</div><div>Via: SIP/2.0/UDP [FS1 IP];rport;branch=z9hG4bKXt9pr4er2H2Np</div><div>Max-Forwards: 69</div><div>From: "[Source Number]" <sip:[Source Number]@[FS1 IP]>;tag=7ZDB1S0pFpSja</div><div>To: <<a href="sip:9499@">sip:9499@</a>[AddPac IP]></div><div>Call-ID: 765f438b-26e6-122d-3490-51d73cb8d94c</div><div>CSeq: 120957528 INVITE</div><div>Contact: <<a href="sip:mod_sofia@">sip:mod_sofia@</a>[FS1 IP]:5060></div><div>User-Agent: Proxy 1.1</div><div>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH</div><div>Supported: timer, precondition, path, replaces</div><div>Allow-Events: talk, presence, dialog, 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: 228</div><div>Remote-Party-ID: <[Source Number]></div><div><br></div><div>v=0</div><div>o=root 16330 16330 IN IP4 [FS2 IP]</div><div>s=session</div><div>c=IN IP4 [FS1 IP]</div><div>t=0 0</div><div>m=audio 27382 RTP/AVP 0 101</div><div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-16</div><div>a=silenceSupp:off - - - -</div><div>a=ptime:20</div></div><div><br></div><div>Packet 3</div><div>=======</div><div><div>SIP/2.0 200 OK</div><div>Via: SIP/2.0/UDP [FS1 IP];rport;branch=z9hG4bKXt9pr4er2H2Np</div><div>From: "[Source Number]" <sip:[Source Number]@[FS1 IP]>;tag=7ZDB1S0pFpSja</div><div>To: <<a href="sip:9499@">sip:9499@</a>[AddPac IP]>;tag=a34a0003a4</div><div>Call-ID: 765f438b-26e6-122d-3490-51d73cb8d94c</div><div>CSeq: 120957528 INVITE</div><div>Supported: timer, replaces, early-session</div><div>User-Agent: AddPac SIP Gateway</div><div>Contact: <a href="sip:9499@">sip:9499@</a>[AddPac IP]</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO</div><div>Content-Type: application/sdp</div><div>Content-Length: 235</div><div><br></div><div>v=0</div><div>o=9499 1254144425 1254144425 IN IP4 [AddPac IP]</div><div>s=AddPac Gateway SDP</div><div>c=IN IP4 [AddPac IP]</div><div>t=1254144425 0</div><div>m=audio 23004 RTP/AVP 0 101</div><div>a=ptime:20</div><div>a=rtpmap:0 PCMU/8000/3</div><div>a=rtpmap:101 telephone-event/8000/1</div><div>a=fmtp:101 0-15 </div><div><br></div><div><br></div><div><div>Packet 4</div><div>=======</div><div><br></div></div><div><div>SIP/2.0 200 OK</div><div>Via: SIP/2.0/UDP [FS2 IP]:5060;branch=z9hG4bK60c74dbf;rport=5060</div><div>From: "[Source Number]" <sip:[Source Number]@[FS2 IP]>;tag=as3bd06925</div><div>To: <<a href="sip:9499@">sip:9499@</a>[FS1 IP]>;tag=6pmjZyFKjD3Ze</div><div>Call-ID: 14b07c2014d3a0aa7719efa03d979eed@[FS2 IP]</div><div>CSeq: 102 INVITE</div><div>Contact: <<a href="sip:mod_sofia@">sip:mod_sofia@</a>[FS1 IP]:5060;transport=udp></div><div>User-Agent: Proxy 1.1</div><div>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH</div><div>Supported: timer, precondition, path, replaces</div><div>Allow-Events: talk, presence, dialog, 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: 221</div><div><br></div><div>v=0</div><div>o=9499 1254144425 1254144425 IN IP4 [AddPac IP]</div><div>s=AddPac Gateway SDP</div><div>c=IN IP4 [FS1 IP]</div><div>t=1254144425 0</div><div>m=audio 0 RTP/AVP <font class="Apple-style-span" size="3"><span class="Apple-style-span" style="font-size: 12px; "><b>96</b></span></font> 101</div><div><font class="Apple-style-span" size="3"><span class="Apple-style-span" style="font-size: 12px; "><b>a=rtpmap:96 PCMU/8000/3</b></span></font></div><div>a=rtpmap:101 telephone-event/8000/1</div><div>a=fmtp:101 0-15</div><div>ACK <a href="sip:mod_sofia@">sip:mod_sofia@</a>[FS1 IP]:5060;transport=udp SIP/2.0</div><div>Via: SIP/2.0/UDP [FS2 IP]:5060;branch=z9hG4bK56269b6a;rport</div><div>From: "[Source Number]" <sip:[Source Number]@[FS2 IP]>;tag=as3bd06925</div><div>To: <<a href="sip:9499@">sip:9499@</a>[FS1 IP]>;tag=6pmjZyFKjD3Ze</div><div>Contact: <sip:[Source Number]@[FS2 IP]></div><div>Call-ID: 14b07c2014d3a0aa7719efa03d979eed@[FS2 IP]</div><div>CSeq: 102 ACK</div><div>User-Agent: Legacy</div><div>Max-Forwards: 70</div><div>Remote-Party-ID: "[Source Number]" <sip:[Source Number]@[FS2 IP]>;privacy=off;screen=no</div><div>Content-Length: 0</div></div><div><br></div><div>I've tried the same test with different codecs, and it is always messing with the payload.</div><div><br></div><div>Thanks in advance</div><div>Regards</div><div>M</div><div><br></div></div></body></html>