Video/Audio Issues Between Clients on Different Mobile Networks (Airtel/Jio)

[soumya26-pi]

Hello,

I’m encountering a problem with video and audio when making Asterisk calls between clients using different mobile network providers (Airtel and Jio).

Problem Description:

Client 1 - uses Airtel mobile data.
Client 2 - uses Jio mobile data.

• When both clients are on different networks (Airtel and Jio), the video stream from one of the clients is either not coming through or there are significant issues with video quality.

• However, if both clients are on the same mobile network (both on Airtel or both on Jio), the video works perfectly fine without any issues.

• What could be causing the video to fail between clients on different mobile networks? Could this be related to NAT traversal, codec negotiation, or something specific to how mobile networks handle RTP video packets?

• Are there any specific configurations I should look into to handle video streams between clients on different mobile networks?

Any insights or suggestions would be greatly appreciated!

[Pooh]

On Sunday 15 September 2024 at 21:53:31, soumya26-pi via Asterisk Community
wrote:

I’m encountering a problem with video and audio when making Asterisk calls
between clients using different mobile network providers (Airtel and Jio).

How are these calls coming in to Asterisk? Are these VoIP (SIP) calls end-to-
end, or do these mobile service providers support video over GSM?

• What could be causing the video to fail between clients on different
  mobile networks?

I would start by looking at the SIP Invites and specifically the SDP
negotiations between Asterisk and the two endpoints.

See what is different between [both clients on same network] and [clients on
different networks].

That should start pointing to where a solution might lie.

Antony,

–
I wasn’t sure about having a beard at first, but then it grew on me.

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                                                     please *don't* CC me.

[soumya26-pi]

It is a sip call(audio and Video) and I’m experiencing one-way audio and video in a SIP call between two endpoints (8003 and 2001udp). Specifically, 8003 can see and hear everything from 2001udp, but 2001udp cannot receive audio or video from 8003. The media flow from 2001udp to 8003 is working fine, but the reverse (from 8003 to 2001udp) is not.

Both endpoints are registered via PJSIP on Asterisk. Below are the configuration details and the troubleshooting steps I’ve taken so far, but the problem persists.

Environment:

• Asterisk version: 18.23
• Transport: UDP for both endpoints (8003, 2001udp), WebRTC for 2001webrtc
• Network setup:
  • Asterisk is running on an EC2 instance
  • 8003 is behind NAT
  • 2001udp is also behind NAT

Symptoms:

• 8003 receives both audio and video from 2001udp, but 2001udp does not receive any media from 8003.
• Both calls show COMPLETED in sngrep (screenshots provided below), but media is only flowing in one direction.
• The SDP exchange between the endpoints seems normal, but it looks like RTP is either not arriving or not being processed correctly on the 2001udp side.

What I’ve Done:

1. PJSIP Configuration:

• NAT traversal settings have been enabled. Below is my configuration for both endpoints:Transport configuration:

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
external_media_address= Public IP of Server
external_signaling_address=Public IP of Server
local_net=Local_net

2.Endpoint for 2001udp*:

[2001udp]
type=endpoint
aors=2001udp
auth=2001udp
context=from-internal
disallow=all
allow=opus,ulaw,g722,vp8,vp9,h264
transport=transport-udp
direct_media=no             ; Asterisk stays in media path
force_rport=yes             ; For NAT traversal
rewrite_contact=yes         ; Rewrites SIP headers for NAT