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<p class="MsoNormal"><span style="color:#1F497D">Hi All,<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">How can tell Freeswitch to set the mark bit when going from early media to media? I am having a one way audio issue for the 1<sup>st</sup> 20 seconds with calls originating from a Cyberdata door box. FS
</span><span style="color:#1F497D">Version 1.2.21</span><span style="color:#1F497D"><o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">Below is my response from Cyberdata Support.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">Hello Moishe,<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">I really appreciate your time working with me today attempting to get the intercom registered! I know I said I would just touch bases with you tomorrow, but I think I have resolved the issue on my end after
talking to our engineers and also figured out what is happening with the intercoms experiencing delayed audio at the intercom’s speaker during calls.
<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">Recently, we resolved a similar issue with a FreeSWITCH provider who reported the same symptoms with their intercoms. Their FreeSWITCH server also transitions from early media (where the ringback tone plays
to the calling device) to live audio just like your FreeSWITCH implementation.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">We learned that our device wasn’t properly identifying the Mark bit in the RTP stream and synchronizing the audio stream. A fix was implemented and the audio delay was resolved with v9.1.1b01.
<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">In your FreeSWITCH implementation, we are seeing the same symptoms but in your case FreeSWITCH does not set the Mark bit in the live audio RTP stream. It does not change the SSRC either. Instead, the timestamp
jumps which causes the intercom to start recording missed timestamps until it hits a threshold after about 20 seconds, and after this point it resynchronizes the audio stream and the intercom starts playing it out of the speaker.
<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">In the attached trace you provided to us, frame 198 shows the transition from early media to live audio. The marker bit is not set. I also included a .jpg highlighting frame 184 and 198 for your reference.
You will see the timestamp jumps but the SSRC/sequence numbers don’t change and the mark bit is not set to indicate a new stream.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">Based on this information, we are recommending you consult with your FreeSWITCH engineers and advise setting the Mark bit in the RTP stream when transitioning from early media to live audio since the SSRC and
sequence numbers of the audio stream do not change. Any calls with the Mark bit set in the new audio stream when switching from early media to live audio should play immediately without delay. However, for calls with new audio streams without a Mark bit
set, audio will be delayed until this is resolved in FreeSWITCH. <o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">I hope this information is helpful, Moishe! Could you please keep us posted?<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">We would be happy to get on a conference call with you if it will help. We know of another FreeSWITCH reseller in your area who had a similar issue so you are not alone.
</span><span style="font-family:Wingdings;color:#1F497D">J</span><span style="color:#1F497D"><o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">Many thanks,<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">Christina<o:p></o:p></span></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:7.5pt;font-family:"Verdana","sans-serif";color:#333399">Thanks,<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:7.5pt;font-family:"Verdana","sans-serif";color:#333399"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="font-size:7.5pt;font-family:"Verdana","sans-serif";color:#333399">Moishe Grunstein<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:7.5pt;font-family:"Verdana","sans-serif";color:#333399">Tornado Computer Systems, Inc.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:7.5pt;font-family:"Verdana","sans-serif";color:#333399">212.400.7650 888.IPPBX.US<br>
</span><b><span style="font-size:7.5pt;font-family:"Verdana","sans-serif";color:#76923C">Service Request Email:
<a href="mailto:support@nysolutions.com"><span style="color:blue">support@nysolutions.com</span></a>
</span></b><span style="font-size:10.0pt;font-family:"Arial","sans-serif";color:#333399"><o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size:7.5pt;font-family:"Verdana","sans-serif";color:#333399">Polycom Certified VAR<br>
Microsoft Small Business Specialist, Cisco SMB Select Certified</span><span style="color:#333399"><o:p></o:p></span></p>
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