Pjproject - Sending STUN Request : Network is unreachable

Asterisk Asterisk WebRTC
1

hello all,

I am using pjsip for video call in asterisk version 16 between the endpoints in different Pc’s.

The video call is working but somewhat i am getting the subjected error in asterisk CLI , after that the call ends… i am using google stun in webrtc.plz help out fot thiz issue:

[May 4 18:38:30] ERROR[1232]: pjproject: <?>: icess0x374c708 …Error sending STUN request: Network is unreachable
[May 4 18:38:30] ERROR[1232]: pjproject: <?>: icess0x3766918 …Error sending STUN request: Network is unreachable
[May 4 18:38:30] ERROR[1232]: pjproject: <?>: icess0x7f888c0552e8 …Error sending STUN request: Network is unreachable
[May 4 18:38:30] ERROR[1232]: pjproject: <?>: icess0x7f888c06f4f8 …Error sending STUN request: Network is unreachable
[May 4 18:38:30] ERROR[1232]: pjproject: <?>: icess0x374c708 …Error sending STUN request: Network is unreachable
[May 4 18:38:30] ERROR[1232]: pjproject: <?>: icess0x3766918 …Error sending STUN request: Network is unreachable
[May 4 18:38:30] ERROR[1232]: pjproject: <?>: icess0x7f888c0552e8 …Error sending STUN request: Network is unreachable
[May 4 18:38:30] ERROR[1232]: pjproject: <?>: icess0x7f888c06f4f8 …Error sending STUN request: Network is unreachable

2

This occurs due to usage of IPv6 link local addresses as ICE candidates. If media is flowing, there’s nothing to do there. You can ignore them.

3

Hi jcolp,

can u give me some clarification the above error does not affect the video quality ri8?

because i am using h264,vp8 codecs for video call…Video is working but still the quality is not satisfied. what can i do for that

4

The error would have absolutely no impact on video quality. You need to state what about video quality isn’t good.

5

I am getting this hits in asterisk cli

[May 4 20:45:24] WARNING[29888]: pjproject: <?> sdp.c .Error adding media attribute, attribute is ignored: Too many objects of the specified type (PJ_ETOOMANY)

Is there any way to control my bandwidth in video calling through webrtc?

Below is my pjsip endpoint configuration:

[xxxx]

type=endpoint
aors=xxxx
auth=xxxx
use_avpf=yes
media_encryption=dtls
dtls_ca_file=/etc/asterisk/keys/asterisk.pem
dtls_cert_file=/etc/asterisk/keys/asterisk.pem
dtls_verify=fingerprint
dtls_setup=actpass
ice_support=yes
media_use_received_transport=yes
rtcp_mux=yes
context=from-internal
disallow=all
allow=alaw,ulaw,g729,h264,vp8
direct_media=no
ice_support=yes
tos_video=af41
cos_video=4
callerid=xxxx

[xxxx]
type=auth
auth_type=userpass
password=xxxx
username=xxxx
[xxxx]
type=aor
max_contacts=7

6

There is no ability within Asterisk to control the bandwidth explicitly. It’s done client side. Are you using bundled PJSIP? What version of Asterisk is this? You should also be using the “webrtc” option as it enables various WebRTC functionality for better experience.

7

I am using asterisk version 16.2.1 so it would default configured with bundled PJSIP.

How can i use “webrtc” option?

8

I do not recommend using old versions of Asterisk for WebRTC. We continue to add improvements. As for the webrtc option it’s set on the endpoint[1].

[1] https://blogs.asterisk.org/2017/09/06/upcoming-webrtc-improvements-asterisk-15/
[2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Configuration_res_pjsip#Asterisk16Configuration_res_pjsip-endpoint_webrtc