[asterisk-bugs] [JIRA] (ASTERISK-26729) Asterisk behind NAT not sending audio according to SDP

[Jared Hull (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26729?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=235821#comment-235821 ] 

Jared Hull edited comment on ASTERISK-26729 at 3/14/17 3:03 PM:
----------------------------------------------------------------

Yea I was hoping it would; this is what we have on our endpoints and trunk:
rtp_keepalive=1
rtp_symmetric=yes
force_rport=yes
direct_media=no

This results in working RTP for dialing to an extension, but sending a reINVITE does not punch through NAT for video.
exten => 142,1,Dial(PJSIP/143) ;This didn't used to work until an upgrade to 14.3 (not sure if that was the cause)

This results in ONLY a 100 Trying response (nothing else after, very strange) to the INVITE for an echo.
exten => 142,1,echo()

These result in working RTP for an echo, but sending a reINVITE does not punch through NAT for video.
exten => 142,1,answer()
same => n,echo()
exten => 143,1,Playback(silence/1)
same => n,echo()


was (Author: fortytwo):
Yea I was hoping it would; this is what we have on our endpoints and trunk:
rtp_keepalive=1
rtp_symmetric=yes
force_rport=yes
direct_media=no

This results in working RTP for dialing to an extension, but sending a reINVITE does not punch through NAT for video.
exten => 142,1,Dial(PJSIP/143) ;This didn't used to work until an upgrad to 14.3 (not sure if that was the cause)

This results in ONLY a 100 Trying response (nothing else after, very strange) to the INVITE for an echo.
exten => 142,1,echo()

These result in working RTP for an echo, but sending a reINVITE does not punch through NAT for video.
exten => 142,1,answer()
same => n,echo()
exten => 143,1,Playback(silence/1)
same => n,echo()

> Asterisk behind NAT not sending audio according to SDP
> ------------------------------------------------------
>
>                 Key: ASTERISK-26729
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26729
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 13.13.1
>         Environment: x64 CentOS
>            Reporter: Luke Escude
>            Assignee: Unassigned
>         Attachments: call1.pcap, Call1.txt, echo1.pcap, Echo1.txt, Video2.txt, video.pcap, Video.txt
>
>
> We are running Asterisk 13.13.1 with PJSIP.
> The setup is as follows:
> The Asterisk boxes are all virtualized behind NAT - let's say their address space is 172.x.x.x. The router that controls that nat is IP address 10.0.4.1. Kamailio/RTPProxy is running on 10.0.1.1.
> Kamailio essentially looks like a "public" IP to the asterisk boxes - we have them REGISTERING to kamailio to IP address 10.0.1.1 (which is routable of course). That way, Kamailio knows how to get to the Asterisks via their "public IP" (10.0.4.1) and NAT port (probably like 16875 or whatever).
> SIP communication happens wonderfully with this setup.
> However, when Kamailio sends an SDP containing its address 10.0.1.1 and an RTP address, it seems like Asterisk doesn't want to send audio to that address. In order for a new NAT port to open, Asterisk has to "talk" first.
> Regardless of our pjsip configuration, we cannot get Asterisk to behave properly, even by following the wiki and setting the external signaling/media addresses.



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