Outgoing SIP calls drop after 32s

And if you use pulse?

What is pulse and how to use it?

My system has a package called pulseaudio that is installed, I have read the wikipedia page on pulseaudio but I still have no clue what the suggestion is.

Is it to activate some option when using Jami? Or trying something else? Any link saying what to do?

In media settings, for the backend you have a choice between alsa & pulseaudio (if present). So my point is to know if it's related to alsa or not, that's why I'm asking if you can try to use pulseaudio instead

I tried, it is exactly the same. Do you want logs of it too?

yeah at least you must not see any "underrun" as alsa is not used, so it can show the issue more clearly

Here it is.

dring6.log

@mzhang can you take a look at this?

[0m[22;36m[1615901643.000|19539|sipcall.cpp       :1389 ] [0m[01;33m[call:3932270934633767] medias changed
[0m[22;36m[1615901643.000|19536|sipcall.cpp       :1403 ] [0m[01;33m[call:3932270934633767] no remote ICE for medias
[0m[22;36m[1615901643.000|19536|sipcall.cpp       :1329 ] [0m[call:3932270934633767] stopping all medias
[0m[22;36m[1615901643.100|19754|media_decoder.cpp :496  ] [0mDecoding audio using PCM A-law / G.711 A-law (pcm_alaw)

I think it may be related to https://git.jami.net/savoirfairelinux/ring-daemon/-/issues/201

@Avron last question. Do you know what codecs are supported by your SIP provider? Only G711?

assigned to @mzhang

added SIP label

Just found out that I can configure codecs priority order, with the following options:

g711
g729, g711
g711, g729
g722, g729, g711
g722, g711, g729

Now the configuration is g711, g729

EDIT: a note says g722 will only be used between SIP lines of my provider

One more info, the last line of the log when the call was just started is: [1615901647.824|19762|media_decoder.cpp :506 ] Not using hardware decoding for pcm_alaw

yeah we do support PCMA, but we don't support G711 nor G729 for now. (cf in the account settings => advanced settings => Audio codecs & this ticket https://git.jami.net/savoirfairelinux/ring-daemon/-/issues/201).

So medias will not negotiate and the SIP will timeout.

Really? In Jami parameters I see pcma and pcmu and from what I read on wikipedia I thought these were the two flavours of g711.

Besides, the call works for 32s and incoming calls are not dropped.

Do you mean Jami does not support my provider's parameters now so there is nothing to do?

HI @Avron

Do you mind using Wireshark to capture the sip packages?

By that, I can have a clearer understanding.

But, according to the log, the connection is cut because there is no data transmitting between UAS and UAC.

Thanks

nop I don't mean this. The call shouldn't be dropped, was just exploring. @mzhang will take a look

maybe related to this issue ring-client-gnome#1259

@Avron can this issue be closed as fixed as:

In recent years, we have focused on peer-to-peer communication and had neglected Jami's ability to work as a SIP phone. This gap is now closed with MALOYA which offers better SIP protocol support, allowing Jami users to use SIP-compatible telephone systems and service providers. This feature brings the ability to place and receive calls with regular phone numbers. Please try out MALOYA’s SIP phone feature, and let us know about your experience! Your feedback will be greatly appreciated and immensely useful in helping us further improve Jami’s SIP phone features.

https://jami.net/maloya-a-new-version-of-jami/

Thank you

I have https://dl.jami.net/nightly/ubuntu_18.04/ jami main in my apt sources, does that mean I have this new version or not?

If yes, I tried again, the call is dropped after 32s exactly, same like before. I am sorry I did not provide wireshark logs yet, I will soon have time to do it if useful.

So far, for SIP, I am using Linphone which works without problem, hence why this is not a critical issue for me.

I am encountering the same issue. I am using Maloya.

I cannot provide wireshark logs, I do not have the free time to download / learn a whole new application.

Hi. I can confirm that I have a SIP 408 with the latest Jami version using OVH.com SIP line

I have contacted the support engineers at OVH and they says it's the client that doesn't answer a request from the server.

The same line on the same host is working using zoiper

Hi, May be it's already the case, but just make sure that "Auto Registration After Expired" is enabled.

Hi, May be it's already the case, but just make sure that "Auto Registration After Expired" is enabled.

It is enabled for me. I have flipped pretty much every switch in the settings, and nothing (that still allows me to remain connected) has worked.

As a lark I downloaded jami via flatpak and tried to see if that changed anything. Nope. Still disconnects after 32 seconds exactly.

I think The relevant logs that print out right at the disconnect are:

[xxx.xxx|    x|xxx.xxx   :xxx  ] username = , server = xxx.xxx.xxx.xxx, from = xxx.xxx.xxx.xxx
[xxx.xxx|    x|xxx.xxx   :xxx  ] [call:xxx] INVITE@xxx state changed to 6 (DISCONNCTD): cause=408, tsx@xxx status 408 (Request Timeout)

very heavily redacted, because if I don't recognize a string I have to assume it could de-anonimise me. if it would help (and if it wont de-anon me) I can add more details.

Using linphone "works" for me, in that I can use it to make and receive phone calls, but it sucks at pretty much every other level. the UI sucks (I cant tell which sip account is being called, and I have many) and it breaks my system consistently (every time i open the settings gnome crashes, sometimes rebooting gracefully but other times forcing me to re log in)

unassigned @mzhang

We tested with OVH accounts and I can confirm that we reproduce the issue, the call ends after 32s. We will work on it soon and will this issue when we have more.