<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:'times new roman', 'new york', times, serif;font-size:12pt;color:black;"><div>Hi ,</div><div><br></div><div>I am working with the latest revision of Freeswitch and I am having one way audio problem using SILK/8000 codec on B-leg of the call after Session Refresh or on HOLD. This is due to the fact that when FS sends an INVITE out to B-leg it sends 98 in its SDP offer and B-leg responds back with same code but with payload type "120"</div><div><br></div><div>[ Offer ]</div><div><div>v=0</div><div> o=FreeSWITCH 1303773966 1303773967 IN IP4 192.168.2.10</div><div> s=FreeSWITCH</div><div> c=IN IP4 192.168.2.10</div><div> t=0 0</div><div> m=audio 33766 RTP/AVP 98 9 101</div><div> a=rtpmap:98 SILK/8000</div><div> a=fmtp:98
useinbandfec=1;usedtx=0</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><br></div><div>[ Answer ]</div><div><br></div><div><div> v=0</div><div> o=- 3512815772 3512815773 IN IP4 192.168.4.121</div><div> s=pjmedia</div><div> c=IN IP4 192.168.4.121</div><div> t=0 0</div><div> a=X-nat:0</div><div> m=audio 4002 RTP/AVP 120 96</div><div> a=rtcp:4003 IN IP4 192.168.4.121</div><div> a=rtpmap:120 silk/8000</div><div> a=fmtp:120 useinbandfec=1;usedtx=0</div><div> a=sendrecv</div><div> a=rtpmap:96 telephone-event/8000</div><div> a=fmtp:96 0-15</div></div><div><br></div><div>Freeswitch handles this properly on the initial offer/answer as its using this patch
(<span class="Apple-style-span" style="font-family: 'Lucida Grande'; font-size: 12px; font-weight: bold; -webkit-border-horizontal-spacing: 2px; -webkit-border-vertical-spacing: 2px; ">tell rtp stack about what remote payload type to expect when the receiving end follows the stupid SHOULD as WONT and sends a different dynamic payload number than the one in the offer</span>)</div><div><br></div><div>After one min into the call because Session Timers are enabled Freeswitch sends a Session Refresh with payload type now setting to "120" </div><div><br></div><div>[Refresh Offer]</div><div><div>v=0</div><div> o=FreeSWITCH 1303773966 1303773967 IN IP4 192.168.2.10</div><div> s=FreeSWITCH</div><div> c=IN IP4 192.168.2.10</div><div> t=0 0</div><div> m=audio 33766 RTP/AVP 120 96 9</div><div> a=rtpmap:120 SILK/8000</div><div> a=fmtp:120
useinbandfec=1;usedtx=0</div><div> a=rtpmap:96 telephone-event/8000</div><div> a=fmtp:96 0-16</div><div> a=silenceSupp:off - - - -</div><div> a=ptime:20</div></div><div><br></div><div>The remote end then starts sending RTP packets with payload number "120" in its RTP header and FS stops processing these packets and as a result is resulting in one-way audio issue.</div><div><br></div><div>Any help is appreciated.</div><div><br></div><div>Thanks in advance</div><div>Regards</div><div>Jyotshna</div><div><br></div><div><br></div><div>P.S : The issue starts even when the remote end presses" HOLD" as it sends INVITE on hold with "120" and FS responds back with "120" in its answer.</div><div><br></div><div><br></div><div><br></div><div style="position:fixed"></div>
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