[asterisk-users] Asterisk behind NAT Early Media Video

Wed Apr 11 02:11:13 CDT 2018

I did a quick check between what I have set and your settings below.

You can try the following and see if it helps

In your endpoint:
bind_rtp_to_media_address=yes

With best regards

Florian Floimair
Innovation - Software-Development -  VoIP & DevOps

COMMEND INTERNATIONAL GMBH
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Tel: +43-662-85 62 25
Fax: +43-662-85 62 26
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Von: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] Im Auftrag von Benjamin Marty
Gesendet: Mittwoch, 11. April 2018 08:55
An: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>
Betreff: Re: [asterisk-users] Asterisk behind NAT Early Media Video

I think I found the root cause. The H264 Early Media video is received successfully on the Asterisk Server. It also seems to get processed. But it's send to the private IP of the receipent SIP phone.
For clarification:
178.82.XX.XX is my Public IP of my Internet access. Both phones use this as Public IP via standard Source NAT.
159.89.XX.XX is the IP of the Asterisk Server. For this test I used a Server without Destination NAT. So the eth0 interface has this IP.
Packet capture:
No.     Time                          Source                Destination           Protocol Length Info
    141 2018-04-11 06:40:03.306561    178.82.XX.XX          159.89.XX.XX        H264     64     PT=H264, SSRC=0x3CB1E12D, Seq=19561, Time=319121408 SPS

Frame 141: 64 bytes on wire (512 bits), 64 bytes captured (512 bits)
Ethernet II, Src: JuniperN_34:67:f0 (40:a6:77:34:67:f0), Dst: da:81:42:3d:d0:e7 (da:81:42:3d:d0:e7)
Internet Protocol Version 4, Src: 178.82.169.0, Dst: 159.89.104.193
User Datagram Protocol, Src Port: 5006, Dst Port: 13182
Real-Time Transport Protocol
H.264

No.     Time                          Source                Destination           Protocol Length Info
    142 2018-04-11 06:40:03.306682    159.89.XX.XX        192.168.XX.XX         H264     64     PT=H264, SSRC=0x5EE97C55, Seq=30572, Time=319121408 SPS

Frame 142: 64 bytes on wire (512 bits), 64 bytes captured (512 bits)
Ethernet II, Src: da:81:42:3d:d0:e7 (da:81:42:3d:d0:e7), Dst: IETF-VRRP-VRID_6e (00:00:5e:00:01:6e)
Internet Protocol Version 4, Src: 159.89.104.193, Dst: 192.168.1.185
User Datagram Protocol, Src Port: 10298, Dst Port: 5022
Real-Time Transport Protocol
H.264
PJSIP.conf:
[7004]
type = endpoint
context = internal
rewrite_contact = yes
direct_media = no
rtp_symmetric = yes
;force_rport = yes
disallow = all
allow = g722, alaw, ulaw, gsm, ilbc, h264
aors = 7004
auth = auth7004

[7004]
type = aor
max_contacts = 2

[auth7004]
type=auth
auth_type=userpass
password=1234
username=7004
extensions.conf:
[internal]
exten => _700X,1,Dial(PJSIP/${EXTEN})

2018-04-10 16:43 GMT+02:00 Benjamin Marty <benjamin.marty at gmail.com<mailto:benjamin.marty at gmail.com>>:
I just noticed, the calling device isn't even sending the early media video stream. It just sends an early media audio stream. Is there propably a change in the signaling needed?
(On another P2P SIP Server the early media video works.)

2018-04-10 12:29 GMT+02:00 Benjamin Marty <benjamin.marty at gmail.com<mailto:benjamin.marty at gmail.com>>:
Hi Florian
I already have the external_media_address set in the PJSIP setup. Also the external_signaling_address is set to the Public IP. If I make a call from an Early Media (video&audio) capable device to an Early Media capable device (also video&audio) the Early Media audio works perfectly. But no video. If I sniff with wireshark on the recipent device I just see G711 (audio) RTP traffic. The h264 RTP traffic is missing before I accept the call. After accepting the call the h264 RTP traffic comes through.
The 183 SIP protocoll comes through. Even Asterisk is noticing it:
-- PJSIP/6002-00000013 is making progress passing it to PJSIP/6001-00000012

I tried both Asterisk 15 with pjsip.conf configuration and Asterisk 13 with sip.conf (chan_sip). In both cases I just put the both case AST_FRAME_VIDEO: statements before the two voice cases, like in your diff and recompiled/reinstalled.
Regards
Benjamin

2018-04-10 9:37 GMT+02:00 Floimair Florian <f.floimair at commend.com<mailto:f.floimair at commend.com>>:
Hi Benjamin!

You're obviously using a similar scenario that I have in place for testing.
I initially had issues with early media (not only video also audio) as well in that scenario. What I had to do was to additionally set

external_media_address=<your external IP>

in pjsip.conf

Also, as I wrote the patch for early-media video I'd be interested in any feedback from it.

With best regards

Florian Floimair
Innovation - Software-Development -  VoIP & DevOps

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
Tel: +43-662-85 62 25
Fax: +43-662-85 62 26
http://www.commend.com<https://linkprotect.cudasvc.com/url?a=http%3a%2f%2fwww.commend.com&c=E,1,3-QFS79bl07XJ1At9-FN042YWg_pIhOoaMJ3B13IzEVsdUP_-SFZDUg5wBrnkEzQgB7TrZRQzaiO0icSJ3UXSJSRnjIVOu0661La-Fj5_q1BczQlPWU_otM,&typo=1>

Security and Communication by Commend

FN 178618z | LG Salzburg

-----Ursprüngliche Nachricht-----
Von: asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com> [mailto:asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com>] Im Auftrag von Joshua Colp
Gesendet: Montag, 9. April 2018 18:15
An: asterisk-users at lists.digium.com<mailto:asterisk-users at lists.digium.com>
Betreff: Re: [asterisk-users] Asterisk behind NAT Early Media Video

On Mon, Apr 9, 2018, at 1:05 PM, Benjamin Marty wrote:
> wohoo, so if I unterstand it correctly with that patch early media
> video works over the Asterisk server? In other words the Asterisk
> server get's able to (process/)forward the early media video stream with that patch?

The patch forwards video while in an early media state before the call is answered and bridged, yes.

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