| Summary: | ASTERISK-25242: PJSIP: No audio when Asterisk inside NAT and endpoints outside NAT - implement functionality similar to chan_sip 'rtpkeepalive'? | ||
| Reporter: | Mark Michelson (mmichelson) | Labels: | |
| Date Opened: | 2015-07-09 13:28:19 | Date Closed: | 2015-07-20 15:52:42 |
| Priority: | Major | Regression? | |
| Status: | Closed/Complete | Components: | |
| Versions: | SVN 13.4.0 | Frequency of Occurrence | |
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | When Asterisk is behind a NAT and the two parties involved in the call are in front of the NAT, it is likely that there will be no audio on the call. The reason is that the NAT will not allow the inbound media to Asterisk through since Asterisk has not punched a hole through the NAT from the inside.
There are workarounds that can be performed in the dialplan, such as playing some silence to one of the parties once the call is answered. This, however, is not very elegant, and it may not seem obvious to an Asterisk deployer if encountered. | ||
| Comments: | By: Mark Michelson (mmichelson) 2015-07-09 13:37:58.827-0500 The proposed solution here for PJSIP is to implement the same thing that chan_sip has for its "rtpkeepalive" option. | ||