| Summary: | ASTERISK-26457: [patch] force_rport,auto_comedia: No NAT detection triggered. | ||||
| Reporter: | Alexander Traud (traud) | Labels: | |||
| Date Opened: | 2016-10-11 07:00:20 | Date Closed: | 2016-10-19 10:09:04 | ||
| Priority: | Major | Regression? | |||
| Status: | Closed/Complete | Components: | Channels/chan_sip/General | ||
| Versions: | 11.23.1 13.11.2 14.0.2 | Frequency of Occurrence | |||
| Related Issues: |
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| Environment: | Attachments: | ( 0) just_auto_comedia.patch | |||
| Description: | *Steps to Reproduce*
# Asterisk with {{nat=force_rport,auto_comedia}} # first VoIP/SIP client (caller) uses public IP addresses in its SDP for example not within a NAT, like IPv6 # second VoIP/SIP client (callee) uses private IP addresses in its SDP for example within a NAT, for example IPv4 without STUN # call is established = signaling via SIP is OK *Expected Results* Media (RTP) should flow, because comedia is enabled as Asterisk detected a NAT towards the callee. Asterisk is sending RTP to public IP addresses. *Actual Results* Media is one way (from callee to caller). Asterisk sends the media of the caller to the address mentioned in the SDP message of the callee. That was a private IP address. Therefore media does not reach the callee. Therefore one-way media. *Workaround* {{nat=auto_force_rport,auto_comedia}} fixed the issue for me, because the related code tests for a NAT in that case. Asterisk should test for NAT, whether {{auto_force_rport}} or {{auto_comedia}} is set. This is done in other calling scenarios within Asterisk already. The attached patch does this for this scenario here as well. | ||||
| Comments: | |||||