ffmpeg: add G726, iLBC codecs (SIP: no audio in call)

I made an internal call via fritz.box and I called myself via the telephone by **600. I heared the ringtone but when I accepted the call, neither of us did hear anything. I'm using Kubuntu 18.04. I guess, these are the important lines in the log:

[1577674451.545|19511|sipcall.cpp       :906  ] [call:3558973192781136] startAllMedia()
[1577674451.545|19511|sip_utils.cpp     :203  ] Registered thread 0x7ff0bdff6dd0 (0x4C36)
[1577674451.546|19511|sdp.cpp           :599  ] Could not find rtpmap attribute
[1577674451.546|19511|sdp.cpp           :599  ] Could not find rtpmap attribute
[1577674451.546|19511|sdp.cpp           :616  ] Could not find codec G726-32
[1577674451.546|19511|sdp.cpp           :616  ] Could not find codec G726-32
[1577674451.546|19511|sdp.cpp           :616  ] Could not find codec G726-40
[1577674451.546|19511|sdp.cpp           :616  ] Could not find codec G726-24
[1577674451.546|19511|sdp.cpp           :616  ] Could not find codec iLBC
[1577674451.546|19511|sdp.cpp           :616  ] Could not find codec telephone-event
[1577674451.546|19511|sipcall.cpp       :956  ] [call:3558973192781136] [SDP:slot#0] Missing remote codec
[1577674451.546|19511|video_rtp_session.cpp:91   ] No socket pair, keyframe request callback not possible

Full log: https://pastebin.com/RtTvcmBY

Kind regards

Indeed ffmpeg is not built with G726 or iLBC.

I move this issue to the related project and rename the issue to be a bit more clear

added media label

assigned to @plespagnol

changed title from No audio on SIP calls to ffmpeg: add G726, iLBC codecs (SIP: no audio in call)

mentioned in issue #201 (closed)

Same problem here on Arch Linux: no audio on incoming calls. However, when I check the output of ffmpeg -codecs then I see

 DEA.L. adpcm_g726           G.726 ADPCM (decoders: g726 ) (encoders: g726 )
 DEA.L. adpcm_g726le         G.726 ADPCM little-endian (decoders: g726le ) (encoders: g726le )
 D.A.L. ilbc                 iLBC (Internet Low Bitrate Codec)

So while my ffmpeg cannot encode iLBC it can at least decode it and certainly encoding/decoding G.726 is available. Still, I hear no sound.

How do I create a log so I can see issues with codecs? Also, is there no codec auto-negotiation going on? Why are these calls even established if the codecs are missing?

My Jami version is 20190813

https://git.jami.net/savoirfairelinux/ring-project/wikis/tutorials/Bug-report-guide#logs

The control channel is created without issues, it's the media socket which is not.

Calling from an internal DECT phone connected to the fritzbox, I get

[1578339834.626|57765|sdp.cpp           :599  ] Could not find rtpmap attribute
[1578339834.626|57765|sdp.cpp           :599  ] Could not find rtpmap attribute
[1578339834.626|57765|sdp.cpp           :599  ] Could not find rtpmap attribute
[1578339834.626|57765|sdp.cpp           :616  ] Could not find codec G726-32
[1578339834.626|57765|sdp.cpp           :616  ] Could not find codec G726-32
[1578339834.626|57765|sdp.cpp           :616  ] Could not find codec G726-40
[1578339834.626|57765|sdp.cpp           :616  ] Could not find codec G726-24
[1578339834.626|57765|sdp.cpp           :616  ] Could not find codec telephone-event
[1578339834.626|57765|sdp.cpp           :616  ] Could not find codec iLBC
[1578339834.626|57765|sdp.cpp           :616  ] Could not find codec PCMA
[1578339834.626|57765|sdp.cpp           :616  ] Could not find codec PCMU

It's the same if I call my fritzbox from an external cell phone. How can it not find PCMA/PCMU?

For an outgoing call I see

[1578339790.016|57768|media_encoder.cpp :497  ] [pcm_alaw] Using 4 threads
[1578339790.016|57768|media_encoder.cpp :538  ] [pcm_alaw] Frame size 160
[1578339790.032|57918|media_decoder.cpp :310  ] Decoding audio using PCM A-law / G.711 A-law (pcm_alaw)

or

[1578340276.422|57768|media_encoder.cpp :497  ] [g722] Using 4 threads
[1578340276.423|57768|media_encoder.cpp :538  ] [g722] Frame size 320
[1578340276.437|58529|media_decoder.cpp :310  ] Decoding audio using G.722 ADPCM (g722)

In jami, all codecs are enabled!

Is the fritzbox to blame in the sense that it doesn't offer or convert to PCMA/PCMU when using a DECT phone?

According to the DECT Monitor of the FritzBox, my DECT phones support G.722 and G.726

I just registered to let you know, that I have the same issue on Manjaro Linux. I made an internal call via fritz.box and I called myself via the telephone by **621. I heared the ringtone but when I accepted the call, neither of us did hear anything. Aufter pause/unpause audio was OK on both sides. It is the same with external calls.

I have the same "Could not find codec" lines in the debug output.

Jami "Together" built on 2020-10-24 06:11:59 UTC

Is there really no progress in 11 month?

Isn't it related to the https://git.ring.cx/savoirfairelinux/ring-project/issues/1126 ?

I have exactly the same behavior as described by Kai Nothrdurft : when I pause/unpause an incoming call, the connection is made.

No issue with an outgoing call.

I agree. It looks like 1126 duplicates this one.

To add something new - incoming calls from an external SIP-provider (Universe Cloud / teleforte-universe.com) through the fritzbox are working without any issues.

By the way, with the Fritz!Box-Family, AVM has the biggest market-share for home-routers in Germany (> 50% says AVM). It's a big player in whole Europe (> 10% market share). So something you should take care of.

closed