Search found 4 matches
- Sat Jan 28, 2017 1:10 am
- Forum: Use Cases
- Topic: Forking incoming RTP to a separate server
- Replies: 2
- Views: 10208
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...
- Thu Jan 26, 2017 9:26 am
- Forum: Use Cases
- Topic: Forking incoming RTP to a separate server
- Replies: 2
- Views: 10208
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...
- Thu Jan 26, 2017 9:09 am
- Forum: Technical Support
- Topic: Audio lost when calling tropo through SipSorcery
- Replies: 2
- Views: 1285
Re: Audio lost when calling tropo through SipSorcery
Thank you, that worked very well.
- Fri Dec 02, 2016 11:32 am
- Forum: Technical Support
- Topic: Audio lost when calling tropo through SipSorcery
- Replies: 2
- Views: 1285
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...