Jigasi answers calls and makes outbound invite calls but there is no audio

Hello,

I set up asterisk 16 and jigasi using a pjsip channel following the example in Working configuration for Jitsi/Jigasi with Asterisk/SIP. I can successfully call into a room and can call out from the room, but there is no audio. I see no messages in the log file that seems to help. I suspected a codec mismatch. My SIP provider supports G711 (alaw) G723 and G739. So configuring my endpoint for alaw successfully allowed me to go through the whole dialplan including the prompts.

Jigasi apparently doesn’t support G711a, G723 and G739. Can I setup the dialplan to do transcoding from alaw to G722 or speex or gsm for example for pjsip snd how basically can I do it? I think chan-sip had channel variables like SIP_CODEC and SIP_CODEC_OUTBOUND to do this, but I prefer to use chan_pjsip.

I noticed that Asterisk 18 has Advanced Codec Negotiation, https://wiki.asterisk.org/wiki/display/AST/PJSIP+Advanced+Codec+Negotiation. Must I use 18?

Any suggestions appreciated.
Thanks
Dennis

Or must I find a sip provider that supports one of the codecs that Jigasi uses?

Dennis

,jigasi supports g711, does your sip provider support latching( waiting to receive audio before sending it)?
If codecs don’t match the call will not be setup at all.

Does g711 need to be explicitly configured in sip-communicator.properties? …Encodings.G711 or G711a or …Encodings.alaw?

Both pcma and pcmu are turned on by default https://github.com/jitsi/jigasi/blob/f020438ffbef9c7d17e0bd4977e3a4ec920eae00/jigasi-home/sip-communicator.properties#L36
You will not see the call connected if there are no matching codecs.
Not hearing normally is a network problem or the server not sending to correct address it it not waits to receive any media before start sending.

So I am sure I am using pcma and the call connects. I checked the sip provider online help and they say that nat=yes should be set in the peers. Looking at https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip#Migratingfromchan_siptores_pjsip-NetworkAddressTranslation(NAT) it indicates to put in the endpoints when using pjsip:

rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes

call connects and still no audio.

Is this nat=force_rport,comedia?

I think in chan_sip nat = yes is obsolete and replaced with nat=force_rport,comedia in a peer. What I sent is what I think is the format for an endpoint in chan_pjsip.

Also I have discovered I also need to:( sip provider web site)

Ensure your network firewall allows traffic from the IP Addresses listed below on all UDP Ports.###### Typical UDP Port Range: 5060 - 5080. Customer premise equipment can be configured on alternate Ports.

Signaling Traffic:
199.180.220.89
199.180.220.91
208.89.104.3

Allow all UDP traffic from Media IPs below

Audio/Media RTP proxies:
45.33.71.83
45.33.70.196
157.230.238.197
45.55.33.77
199.180.223.109

I reviewed all the related configuration as recommended by sip provider. They reviewed their logs at the time I made a call and the say everything works well and tested a call themselves and confirmed the audio between the sip provider and asterisk is working and one of the Audio RTP proxies that I configured is being used.

Seems like the problem may be between asterisk and jigasi. Is there any NAT related configuration that I need to do?

Thanks

Nope, NAT is being handled by the sip side.