SIP provider

This commit brings basic SIP integration as a calls plugin. I could probably keep on adding and refining in my private branch but as it's a larger change and should provide a solid basis to build on why not just have it. This will also make reviewing the upcoming improvements/bugfixes easier in separate MRs.

Current issues/things missing (off the top of my head):

• weird issue with no incoming calls on sip.linphone.org I can place outgoing calls fine-ish (404 answer in registration probe)
• still can't be reached on sip.linphone.org ;(
• Remove a macro hack around call string handling and automatically registering with the server
• Should probably unregister from the server when going into suspend
• Echo cancellation? See !270 (comment 145241)
• No test cases yet (need to think what exactly i want to test anyway - something like creating/tearing down the CallsSipProvider/CallsSipOrigin, maybe something like using one of the sip test servers like https://sip5060.net/test-calls/
• Deps in flatpak are missing
• incoming call: ringtone keeps playing after rejecting/the other side hangs up

Trying it out

So if you're feeling adventurous you can give these patches a spin.

SIP config

You will need to create a config ~/.config/calls/sip-account.cfg with either

[Debug]
Direct=1

for direct connections (f.e. 2 Librem5's in the same network) or

[MyAccount]
User=MyUser
Password=SuperSecretPassword123
Host=mysipserver.org
# Port can be omitted, defaults to 5060 for SIP and 5061 for SIPS
Port=5060
# Protocol must be one of UDP,TCP,TLS
Protocol=UDP

I usually have only 1 account active at a time (uncomment the other ones) but there is nothing in the code preventing having multiple accounts.

Startup

Invoke calls with -p sip parameter.

Calling

Invoke calls with sip:user@host parameter.

changed the description

changed the description

Some hype building measure:

Echo cancellation? Maybe this one is purely a alsa ucm configuration thing?

All you need is to ensure that your PulseAudio streams have filter.want property set to echo-cancel and the rest should be handled automatically.

changed the description

mentioned in issue #229 (closed)

added 28 commits

• f4c3bcee...0af0d48e - 6 commits from branch Librem5:master
• 1dafd7c2 - sip: Initial provider
• 90c660a4 - sip: Origin needs account credentials
• ab150912 - sip: sipify provider with sofia
• 0463c033 - sip: Add media manager and sipify origin
• fcfd3cb7 - meson: add sip_testing option (hack for number handling)
• 552490f3 - sip: initial call handling
• d5f233d5 - sip: origin: fix oper_t reference in sip_callback
• e8081f4a - sip: origin: get address on incoming call
• a9c3cb62 - sip: origin: fix direct connection case
• 2151c9fb - sip: call: rework call state changes
• 418bba8a - sip: origin: emit message on DNS error
• 506fb9b0 - sip: origin: do not use hardcoded ports for RTP
• 37b03db8 - sip: pipeline: bind sockets for RTP
• 2bd8797f - sip: origin: register with SIP server
• 35f5fa04 - sip: go offline when disposing CallsSipOrigin
• 36ab0379 - sip: rework setting SDP
• 21a38768 - sip: allow specifying local port and use IPv6
• 5c3ec84e - sip: Use app name and vcs tag in the user agent
• 6b494059 - sip: media: change default codec to PCMA
• c5a4a88f - sip: pipeline: Add debugging information for used sockets
• 3c721686 - sip: hang up call when contact can't be found (404)
• ed3e7149 - sip: media: improve SDP offer/answer handling

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added 2 commits

• 588eb747 - sip: handle i_outbound 404 errors
• f6e9e3b9 - sip: media: improve SDP offer/answer handling

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added 6 commits

• d79d471a - FINISH ME LATER: ipv4/v6
• e634d175 - sip: whitespace
• 0ae2008a - sip: use ipv4 exclusively for now
• 5ad3b770 - squash with ipv4/6
• 29a10a75 - sip: test using idle callback for register
• 72a493d5 - sip: experiment with parameters

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added 1 commit

• 09514fec - sip: include expire, urn uuid

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added 2 commits

• 9902571c - sip: use ipv4 exclusively for now
• 3e8f734e - sip: improve connection handling by using relevant sofia tags

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the commits mention various hacks, so is this considered WIP?

pipeline fails due to use of deprecated function: https://source.puri.sm/evangelos.tzaras/calls/-/jobs/331722#L2301

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• a58190e7 - application: open sip uri

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• 90c4fcc8 - application: open sip uri

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added 28 commits

• 45dd165c - flatpak: remove libhandy as it is part of the SDK
• 35f45d83 - meson: bump required libhandy-1 version >1.1.90
• ffc3f095 - contact-row: remove deprecated function
• e2870f46 - sip: Initial provider
• ba429203 - sip: Origin needs account credentials
• fcb1380a - sip: sipify provider with sofia
• 00a10027 - sip: Add media manager and sipify origin
• 003946cf - meson: add sip_testing option (hack for number handling)
• ea558df2 - sip: initial call handling
• 1212aeef - sip: origin: fix oper_t reference in sip_callback
• 46515f06 - sip: origin: get address on incoming call
• 630be4b1 - sip: origin: fix direct connection case
• 2225e48c - sip: call: rework call state changes
• 4d329040 - sip: origin: emit message on DNS error
• e1d16f01 - sip: origin: do not use hardcoded ports for RTP
• 32e1fcf5 - sip: pipeline: bind sockets for RTP
• df46c3fd - sip: origin: register with SIP server
• 7c110cdd - sip: go offline when disposing CallsSipOrigin
• 621732d1 - sip: rework setting SDP
• ef916a84 - sip: allow specifying local port and use IPv6
• 14c3e36e - sip: Use app name and vcs tag in the user agent
• 6490f793 - sip: media: change default codec to PCMA
• cbc50136 - sip: pipeline: Add debugging information for used sockets
• 0baf9aaf - sip: handle i_outbound 404 errors
• 6856b2b7 - sip: media: improve SDP offer/answer handling
• 80deac6d - sip: use ipv4 exclusively for now
• 4869543c - sip: improve connection handling by using relevant sofia tags
• 6e007272 - application: open sip uri

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v2: rebased on !270 (merged) currently testing if i can get rid of the nasty hacks..

changed the description

added 26 commits

• 3aebfcc6 - contact-row: remove deprecated function
• 53603dba - sip: Initial provider
• 89680d10 - sip: Origin needs account credentials
• 8d584efe - sip: sipify provider with sofia
• 785f6f1d - sip: Add media manager and sipify origin
• d56839d7 - meson: add sip_testing option (hack for number handling)
• a404a49d - sip: initial call handling
• dad6f04d - sip: origin: fix oper_t reference in sip_callback
• 3f479180 - sip: origin: get address on incoming call
• d3f07839 - sip: origin: fix direct connection case
• 65dc7020 - sip: call: rework call state changes
• 367334e6 - sip: origin: emit message on DNS error
• b5a7d684 - sip: origin: do not use hardcoded ports for RTP
• 2f0b0ff3 - sip: pipeline: bind sockets for RTP
• 2744ac12 - sip: origin: register with SIP server
• e4197e07 - sip: go offline when disposing CallsSipOrigin
• c9702d08 - sip: rework setting SDP
• 0c0bb884 - sip: allow specifying local port and use IPv6
• bb8a39a2 - sip: Use app name and vcs tag in the user agent
• 792c9139 - sip: media: change default codec to PCMA
• da14d341 - sip: pipeline: Add debugging information for used sockets
• 533581f2 - sip: handle i_outbound 404 errors
• 34db4239 - sip: media: improve SDP offer/answer handling
• 131f3fc5 - sip: use ipv4 exclusively for now
• 0e786101 - sip: improve connection handling by using relevant sofia tags
• 26ac4d89 - application: open sip uri

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added 20 commits

• c5cbe0bc - sip: initial call handling
• 9346fd68 - sip: origin: fix oper_t reference in sip_callback
• f28d059a - sip: origin: get address on incoming call
• e4d65a4a - sip: origin: fix direct connection case
• 1f5f0e99 - sip: call: rework call state changes
• 09b51fe6 - sip: origin: emit message on DNS error
• d7b767d0 - sip: origin: do not use hardcoded ports for RTP
• 81f2b216 - sip: pipeline: bind sockets for RTP
• f9ae597e - sip: origin: register with SIP server
• 75d6ca59 - sip: go offline when disposing CallsSipOrigin
• 8ae2edbd - sip: rework setting SDP
• b401d571 - sip: allow specifying local port and use IPv6
• c6876b7d - sip: Use app name and vcs tag in the user agent
• 4a99ed0e - sip: media: change default codec to PCMA
• 554f3248 - sip: pipeline: Add debugging information for used sockets
• 70085c68 - sip: handle i_outbound 404 errors
• 6768f113 - sip: media: improve SDP offer/answer handling
• a670d1c2 - sip: use ipv4 exclusively for now
• e87c8a52 - sip: improve connection handling by using relevant sofia tags
• 7b8a3343 - application: open sip uri

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added 8 commits

• 5f426c17 - sip: Use app name in the user agent
• d274aa4b - sip: media: change default codec to PCMA
• f19a85f5 - sip: pipeline: Add debugging information for used sockets
• 5ceaa03a - sip: handle i_outbound 404 errors
• 9d7848ef - sip: media: improve SDP offer/answer handling
• 436f2ec9 - sip: use ipv4 exclusively for now
• 0b8cc88a - sip: improve connection handling by using relevant sofia tags
• fdfce59a - application: open sip uri

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marked the checklist item Remove a macro hack around call string handling and automatically registering with the server as completed

marked the checklist item still can't be reached on sip.linphone.org ;( as completed

v3: got rid of hacks, can now simply place a call with `gnome-calls sip:user@host", removed vcs tag from user agent

missing: updated flatpak manifest to include sofia-sip

added 22 commits

• a01af6dd - sip: Add media manager and sipify origin
• 613c59ee - sip: initial call handling
• 9ab31f99 - sip: origin: fix oper_t reference in sip_callback
• 45b3efd9 - sip: origin: get address on incoming call
• 0b407f3f - sip: origin: fix direct connection case
• 17149db5 - sip: call: rework call state changes
• 11945a00 - sip: origin: emit message on DNS error
• 3eae2dff - sip: origin: do not use hardcoded ports for RTP
• 652c7772 - sip: pipeline: bind sockets for RTP
• cfd80afd - sip: origin: register with SIP server
• 847e576b - sip: go offline when disposing CallsSipOrigin
• 3e9b1397 - sip: rework setting SDP
• 5bd41c92 - sip: allow specifying local port and use IPv6
• 5e84dbe4 - sip: Use app name in the user agent
• 253a1068 - sip: media: change default codec to PCMA
• 533f2c03 - sip: pipeline: Add debugging information for used sockets
• 123009dd - sip: handle i_outbound 404 errors
• f9552eec - sip: media: improve SDP offer/answer handling
• c3e6151d - sip: use ipv4 exclusively for now
• 5f2a210f - sip: improve connection handling by using relevant sofia tags
• f2150f80 - application: open sip uri
• 0eb9dbb3 - sip: use g_return_if_fail and friends only for public functions

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added 3 commits

• 2e3bb40b - sip: use g_return_if_fail and friends only for public functions
• cd3c510e - squash with 89680d
• e8dc0baa - squash with a01af

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added 1 commit

• b6c9be06 - sip: use g_return_if_fail and friends only for public functions

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added 1 commit

• 626c8310 - sip: use g_return_if_fail and friends only for public functions

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added 2 commits

• 446ec784 - application: open sip uri
• 567911f5 - sip: use g_return_if_fail and friends only for public functions

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added 13 commits

• 2398470e - sip: origin: register with SIP server
• 56a28ff1 - sip: go offline when disposing CallsSipOrigin
• 0fe7bda1 - sip: rework setting SDP
• ff6e41c8 - sip: allow specifying local port and use IPv6
• 0ab45973 - sip: Use app name in the user agent
• 038e2c07 - sip: media: change default codec to PCMA
• 1cf85cf5 - sip: pipeline: Add debugging information for used sockets
• 03afab5f - sip: handle i_outbound 404 errors
• ca7b81ce - sip: media: improve SDP offer/answer handling
• 35aaa5f9 - sip: use ipv4 exclusively for now
• 534dbd08 - sip: improve connection handling by using relevant sofia tags
• 989ad51d - application: open sip uri
• aa12d440 - sip: use g_return_if_fail and friends only for public functions

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v4: reworked sip uri handling,

missing/issue:

• updated flatpak manifest to include sofia-sip
• both when rejecting as well as when the callee cancels, the ringtone keeps on playing :O

resolved all threads

changed the description

added 3 commits

• ba39bf8e - sip: improve connection handling by using relevant sofia tags
• 4cd2e9cb - application: open sip uri
• 05398b4c - sip: use g_return_if_fail and friends only for public functions

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added 3 commits

• 8c343d4f - sip: improve connection handling by using relevant sofia tags
• 0fb259eb - application: open sip uri
• 68081760 - sip: use g_return_if_fail and friends only for public functions

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added 23 commits

• aba1bd76 - sip: Add media manager and sipify origin
• 8930ae52 - sip: initial call handling
• 0f4b78ca - sip: origin: fix oper_t reference in sip_callback
• 0a9f6615 - sip: origin: get address on incoming call
• 0bef5037 - sip: origin: fix direct connection case
• 8ee4813d - sip: call: rework call state changes
• 662d1ee3 - sip: origin: emit message on DNS error
• 032e62b0 - sip: origin: do not use hardcoded ports for RTP
• 2612c5f6 - sip: pipeline: bind sockets for RTP
• a08485f4 - sip: origin: register with SIP server
• 5bf479d1 - sip: go offline when disposing CallsSipOrigin
• 9a8e16a1 - sip: rework setting SDP
• 2301c3f6 - sip: allow specifying local port and use IPv6
• 0df2155e - sip: Use app name in the user agent
• 1425795d - sip: media: change default codec to PCMA
• 5ee96305 - sip: pipeline: Add debugging information for used sockets
• da225252 - sip: handle i_outbound 404 errors
• 89cc0de3 - sip: media: improve SDP offer/answer handling
• 9291b8aa - sip: use ipv4 exclusively for now
• c138ec0d - sip: improve connection handling by using relevant sofia tags
• 2ac0a32b - application: open sip uri
• cc97469f - sip: use g_return_if_fail and friends only for public functions
• bc36047b - sip: improved authentication (maybe)

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added 1 commit

• eb52aa82 - sip: fix infinite ringtone loop

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mentioned in issue #238 (closed)

added 28 commits

• 03d960cc - 1 commit from branch Librem5:master
• 2f7953aa - sip: Initial provider
• 72aa2703 - sip: Origin needs account credentials
• cd74cf46 - sip: sipify provider with sofia
• 8f735a82 - sip: Add media manager and sipify origin
• 8bc440de - sip: initial call handling
• 1e6144cf - sip: origin: fix oper_t reference in sip_callback
• 6622ba6c - sip: origin: get address on incoming call
• 6c7b0b45 - sip: origin: fix direct connection case
• 01cfbf5f - sip: call: rework call state changes
• 779495a4 - sip: origin: emit message on DNS error
• c70ce392 - sip: origin: do not use hardcoded ports for RTP
• aa8a938c - sip: pipeline: bind sockets for RTP
• 75cb08ac - sip: origin: register with SIP server
• 5a5857fb - sip: go offline when disposing CallsSipOrigin
• 551dccf1 - sip: rework setting SDP
• db255663 - sip: allow specifying local port and use IPv6
• 7c360395 - sip: Use app name in the user agent
• 98721785 - sip: media: change default codec to PCMA
• a765a807 - sip: pipeline: Add debugging information for used sockets
• 4b97007e - sip: handle i_outbound 404 errors
• 2feca066 - sip: media: improve SDP offer/answer handling
• aaab5618 - sip: use ipv4 exclusively for now
• 36402111 - sip: improve connection handling by using relevant sofia tags
• e177bca3 - application: open sip uri
• b6c69168 - sip: use g_return_if_fail and friends only for public functions
• d49cf7de - sip: slightly improved authentication
• ec855f8f - sip: fix infinite ringtone loop

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v5: found and fixed #238 (closed) which was also impacting incoming SIP calls (ringtone would only stop playing when you accept an incoming call - in every other case it would happily continue ringing ;) )

missing still: flatpak does not like me

marked the checklist item incoming call: ringtone keeps playing after rejecting/the other side hangs up as completed

changed the description

added 12 commits

• 815b0236 - sip: allow specifying local port and use IPv6
• e60388fc - sip: Use app name in the user agent
• bf4b95cd - sip: media: change default codec to PCMA
• 62271d18 - sip: pipeline: Add debugging information for used sockets
• 9488e3cf - sip: handle i_outbound 404 errors
• 732c34ca - sip: media: improve SDP offer/answer handling
• 28799a28 - sip: use ipv4 exclusively for now
• 9b1a5c5b - sip: improve connection handling by using relevant sofia tags
• e83e8ebe - application: open sip uri
• d2f521df - sip: use g_return_if_fail and friends only for public functions
• a048524b - sip: slightly improved authentication
• a3f82f22 - sip: fix infinite ringtone loop

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added 1 commit

• 8190db53 - sip-media: enable echo cancellation

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v6: added some echo cancellation which i'm currently testing.

flatpak still missing

added 28 commits

• 0e02c5e2 - sip: Initial provider
• ddb70ba3 - sip: Origin needs account credentials
• f61235a4 - sip: sipify provider with sofia
• 35676da0 - sip: Add media manager and sipify origin
• 11075529 - sip: initial call handling
• e4ab52c8 - sip: origin: fix oper_t reference in sip_callback
• f2c05a0f - sip: origin: get address on incoming call
• 404740e8 - sip: origin: fix direct connection case
• de68b250 - sip: call: rework call state changes
• ba877961 - sip: origin: emit message on DNS error
• 22cdf920 - sip: origin: do not use hardcoded ports for RTP
• f16de836 - sip: pipeline: bind sockets for RTP
• 30b779de - sip: origin: register with SIP server
• a35ac7e7 - sip: go offline when disposing CallsSipOrigin
• 2eca7250 - sip: rework setting SDP
• 7242c2c6 - sip: allow specifying local port and use IPv6
• da876f11 - sip: Use app name in the user agent
• 61159e90 - sip: media: change default codec to PCMA
• efed90f7 - sip: pipeline: Add debugging information for used sockets
• b8d42f68 - sip: handle i_outbound 404 errors
• 859270e6 - sip: media: improve SDP offer/answer handling
• 764b12b4 - sip: use ipv4 exclusively for now
• 1afedeaa - sip: improve connection handling by using relevant sofia tags
• 4db81fb5 - application: open sip uri
• 45dab113 - sip: use g_return_if_fail and friends only for public functions
• fe60d841 - sip: slightly improved authentication
• 2d0c924a - sip: fix infinite ringtone loop
• f3eb13e9 - sip-media: enable echo cancellation

Compare with previous version