Search found 4 matches

by novice
Sat Jan 28, 2017 1:10 am
Forum: Use Cases
Topic: Forking incoming RTP to a separate server
Replies: 2
Views: 431

Re: Forking incoming RTP to a separate server

I read the article about SIP one way audio situations https://sipsorcery.wordpress.com/2009/08/05/nat-rtp-and-audio-problems/ and I thought it was exactly the type of situation that I was trying to recreate: RTP audio from the client is sent to an IP address but incoming audio arrives from a differe...
by novice
Thu Jan 26, 2017 9:26 am
Forum: Use Cases
Topic: Forking incoming RTP to a separate server
Replies: 2
Views: 431

Forking incoming RTP to a separate server

Hi, I am considering building a virtual assistant for incoming phone calls but I am very new to SIP. Several people have recommended to use Asterisk/Freeswitch but I am concerned about scalability of the solution. Alternatively, I am considering existing solutions Plivo, Tropo, Twilio for Text To Sp...
by novice
Thu Jan 26, 2017 9:09 am
Forum: Technical Support
Topic: Audio lost when calling tropo through SipSorcery
Replies: 2
Views: 192

Re: Audio lost when calling tropo through SipSorcery

Thank you, that worked very well.
by novice
Fri Dec 02, 2016 11:32 am
Forum: Technical Support
Topic: Audio lost when calling tropo through SipSorcery
Replies: 2
Views: 192

Audio lost when calling tropo through SipSorcery

Hi, I tried the recipe https://www.sipsorcery.com/mainsite/Help/TropoCallback and it all seems to be working properly except that I cannot hear any audio. I do receive it when I call the original @sip.tropo.com address but it gets lost when I call the @sipsorcery account and enter the @sip.tropo.com...