3 Way Call Conference
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Ashish Gautam
Jul 2, 2018, 10:43:26 AM7/2/18
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Hello Team,
How can we implement the 3 Ways Call Conference?
Iñaki Baz Castillo
Jul 2, 2018, 10:48:33 AM7/2/18
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Just do it in your SIP servers backend and it will work. JsSIP does support incoming INVITE without SDP.
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Iñaki Baz Castillo
<i...@aliax.net>
<i...@aliax.net>
Guest
Jul 5, 2018, 9:40:47 AM7/5/18
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Hi. Maybe you solved this problem?
Iñaki Baz Castillo
Jul 5, 2018, 9:45:23 AM7/5/18
to jssip
Which problem? There is no problem here.
El jue., 5 jul. 2018 15:40, Guest <elena....@mifort.org> escribió:
Hi. Maybe you solved this problem?
Guest
Jul 5, 2018, 9:51:13 AM7/5/18
to JsSIP
If i understood correctly, I can make 3 way conference only on backend? On frontend side it is impossible?
четверг, 5 июля 2018 г., 16:45:23 UTC+3 пользователь Iñaki Baz Castillo написал:
четверг, 5 июля 2018 г., 16:45:23 UTC+3 пользователь Iñaki Baz Castillo написал:
Iñaki Baz Castillo
Jul 5, 2018, 9:53:23 AM7/5/18
to jssip
For making a 3 way conference you need backend cooperation, yes, sure.
Leonardo Costa
Jun 7, 2019, 10:31:15 AM6/7/19
to JsSIP
Hello, guys!
Thanks for the great job! I'm using JsSIP 3.3.6 and everything is working fine!
Now I'm trying to implement a "3-Way Conference - Third Party Is Added" as described on RFC 5359 (Session Initiation Protocol Service Examples), page 101.
- RFC 5359: https://tools.ietf.org/html/rfc5359
Is it possible using JsSIP to send a re-INVITE (like F5 as below) with the same Call-ID, changing the Contact and adding the tag isfocus? I was looking into the API docs and I could't find a method to send this re-INVITE... I was looking into classes JsSIP.RTCSession and JsSIP.UA..
Could you point me some direction?
Thanks!
2.10. 3-Way Conference - Third Party Is Added
Alice Bob Carol | INVITE F1 | | |--------------->| | | 180 Ringing F2 | | |<---------------| | | 200 OK F3 | | |<---------------| | | ACK F4 | | |--------------->| | | RTP | | |<==============>| | | INVITE F5 | | |<---------------| | | 200 OK F6 | | |--------------->| | | ACK F7 | | |<---------------| INVITE F8 | | |------------->| | | 180 F9 | | |<-------------| | | 200 OK F10 | | |<-------------| | | ACK F11 | | |------------->| | | RTP | | |<============>| In this scenario, Alice and Bob are in a 2-party call (session) when Bob wishes to add Carol into the conversation. Bob is capable of media mixing in a 3-party call. Bob first sends a re-INVITE to Alice, changing Contact URIs to one that indicates Bob's mixer and acts like a focus. As a result, Bob includes the "isfocus" feature tag [RFC3840] as described in [RFC4579]. Bob then INVITEs Carol using the same Contact URI. Note that Bob could wait to re-INVITE Alice until after Carol has answered. Bob could also put Alice on hold before calling Carol. Message Details F1 INVITE Alice -> Bob INVITE sips:b...@biloxi.example.com SIP/2.0 Via: SIP/2.0/TLS client.atlanta.example.com:5061 ;branch=z9hG4bK74bf9
Max-Forwards: 70 From: Alice <sips:al...@atlanta.example.com>;tag=1234567 To: Bob <sips:b...@biloxi.example.com> Call-ID: 1234...@atlanta.example.com CSeq: 1 INVITE Contact: <sips:al...@client.atlanta.example.com> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: ... v=0 o=alice 2890844526 2890844526 IN IP4 client.atlanta.example.com s= c=IN IP4 client.atlanta.example.com t=0 0 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 F2 180 Ringing Bob -> Alice SIP/2.0 180 Ringing Via: SIP/2.0/TLS client.atlanta.example.com:5061 ;branch=z9hG4bK74bf9 ;received=192.0.2.103 From: Alice <sips:al...@atlanta.example.com>;tag=1234567 To: Bob <sips:b...@biloxi.example.com>;tag=23431 Call-ID: 1234...@atlanta.example.com CSeq: 1 INVITE Contact: <sips:b54g...@biloxi.example.com> Content-Length: 0 F3 200 OK Bob -> Alice SIP/2.0 200 OK Via: SIP/2.0/TLS client.atlanta.example.com:5061 ;branch=z9hG4bK74bf9 ;received=192.0.2.103 From: Alice <sips:al...@atlanta.example.com>;tag=1234567 To: Bob <sips:b...@biloxi.example.com>;tag=23431 Call-ID: 1234...@atlanta.example.com CSeq: 1 INVITE Contact: <sips:b54g...@biloxi.example.com> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFYSupported: replaces, gruu Content-Type: application/sdp Content-Length: ... v=0 o=bob 2890844527 2890844527 IN IP4 client.biloxi.example.com s= c=IN IP4 client.biloxi.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 F4 ACK Alice -> Bob ACK sips:b54g...@biloxi.example.com SIP/2.0 Via: SIP/2.0/TLS client.atlanta.example.com:5061 ;branch=z9hG4bK74bfL Max-Forwards: 70 From: Alice <sips:al...@atlanta.example.com>;tag=1234567 To: Bob <sips:b...@biloxi.example.com>;tag=23431 Call-ID: 1234...@atlanta.example.com CSeq: 1 ACK Content-Length: 0 /* Alice and Bob have established a session. Bob re-INVITEs, changing Contact URIs. */ F5 INVITE Bob -> Alice INVITE sips:al...@client.atlanta.example.com SIP/2.0 Via: SIP/2.0/TLS client.biloxi.example.com:5061 ;branch=z9hG4bKnashds Max-Forwards: 70 From: Bob <sips:b...@biloxi.example.com>;tag=23431 To: Alice <sips:al...@atlanta.example.com>;tag=1234567 Call-ID: 1234...@atlanta.example.com CSeq: 1024 INVITE Contact: <sips:bob-...@client.biloxi.example.com>;isfocus Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Supported: replaces, gruu Content-Length: ... v=0 o=bob 2890844527 2890844528 IN IP4 client.biloxi.example.com s=c=IN IP4 client.biloxi.example.com t=0 0 m=audio 49172 RTP/AVP 0 a=rtpmap:0 PCMU/8000 F6 200 OK Alice -> Bob SIP/2.0 200 OK Via: SIP/2.0/TLS client.biloxi.example.com:5061 ;branch=z9hG4bKnashds7 ;received=192.0.2.113 From: Bob <sips:b...@biloxi.example.com>;tag=23431 To: Alice <sips:al...@atlanta.example.com>;tag=1234567 Call-ID: 1234...@atlanta.example.com CSeq: 1024 INVITE Contact: <sips:al...@client.atlanta.example.com> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: ... v=0 o=alice 2890844526 2890844526 IN IP4 client.atlanta.example.com s= c=IN IP4 client.atlanta.example.com t=0 0 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 F7 ACK Bob -> Alice ACK sips:al...@client.atlanta.example.com SIP/2.0 Via: SIP/2.0/TLS client.biloxi.example.com:5061 ;branch=z9hG4bKnash3G Max-Forwards: 70 From: Bob <sips:b...@biloxi.example.com>;tag=23431 To: Alice <sips:al...@atlanta.example.com>;tag=1234567 Call-ID: 1234...@atlanta.example.com CSeq: 1024 ACK Content-Length: 0 /* Bob calls Carol. */F8 INVITE Bob -> Carol INVITE sips:ca...@chicago.example.com SIP/2.0 Via: SIP/2.0/TLS client.biloxi.example.com:5061 ;branch=z9hG4bKnashJfd Max-Forwards: 70 From: Bob <sips:b...@biloxi.example.com>;tag=8675309 To: Carol <sips:ca...@chicago.example.com> Call-ID: sdjfd...@biloxi.example.com CSeq: 42 INVITE Contact: <sips:bob-...@client.biloxi.example.com>;isfocus Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Supported: replaces, gruu Content-Type: application/sdp Content-Length: ... v=0 o=bob 28908445834 2890844834 IN IP4 client.biloxi.example.com s= c=IN IP4 client.biloxi.example.com t=0 0 m=audio 48174 RTP/AVP 0 a=rtpmap:0 PCMU/8000 F9 180 Ringing Carol -> Bob SIP/2.0 200 OK Via: SIP/2.0/TLS client.biloxi.example.com:5061 ;branch=z9hG4bKnashJfd ;received=192.0.2.113 From: Bob <sips:b...@biloxi.example.com>;tag=8675309 To: Carol <sips:ca...@chicago.example.com>;tag=341313 Call-ID: sdjfd...@biloxi.example.com CSeq: 42 INVITE Contact: <sips:ca...@client.chicago.example.com> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Supported: replaces Content-Length: 0 F10 200 OK Carol -> Bob SIP/2.0 200 OK Via: SIP/2.0/TLS client.biloxi.example.com:5061 ;branch=z9hG4bKnashJfd ;received=192.0.2.113 From: Bob <sips:b...@biloxi.example.com>;tag=8675309To: Carol <sips:ca...@chicago.example.com>;tag=341313 Call-ID: sdjfd...@biloxi.example.com CSeq: 42 INVITE Contact: <sips:ca...@client.chicago.example.com> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: ... v=0 o=carol 2890844922 2890844922 IN IP4 client.chicago.example.com s= c=IN IP4 client.chicago.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 F11 ACK Bob -> Carol ACK sips:ca...@client.chicago.example.com SIP/2.0 Via: SIP/2.0/TLS client.biloxi.example.com:5061 ;branch=z9hG4bKnash431 Max-Forwards: 70 From: Bob <sips:b...@biloxi.example.com>;tag=8675309 To: Carol <sips:ca...@chicago.example.com>;tag=341313 Call-ID: sdjfd...@biloxi.example.com CSeq: 42 ACK Content-Length: 0 /* Bob's mixer now mixes media from both Alice and Carol to create the 3-way conference. */
Em quinta-feira, 5 de julho de 2018 10:53:23 UTC-3, Iñaki Baz Castillo escreveu:
For making a 3 way conference you need backend cooperation, yes, sure.
On Thu, 5 Jul 2018 at 15:51, Guest <elena....@mifort.org> wrote:
>
> If i understood correctly, I can make 3 way conference only on backend? On frontend side it is impossible?
>
> четверг, 5 июля 2018 г., 16:45:23 UTC+3 пользователь Iñaki Baz Castillo написал:
>>
>> Which problem? There is no problem here.
>>
>> El jue., 5 jul. 2018 15:40, Guest <elena....@mifort.org> escribió:
>>>
>>> Hi. Maybe you solved this problem?
>>>
>>>>
>>>>
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>
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José Luis Millán
Jun 7, 2019, 11:50:24 AM6/7/19
to js...@googlegroups.com
Do you just need to add the isfocus parameter in the Contact header field?
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José Luis Millán
Leonardo Costa
Jun 7, 2019, 1:09:20 PM6/7/19
to JsSIP
Hello, José!
In fact I want to follow the 3-way conference example flow as described on the RFC to make it work! I already have one conversation call (session) stablished between 2 users (Bob and Alice)! Following the RFC flow, the next step is to Bob send a re-INVITE to Alice (step F5), changing the Contatct URI and adding a the tag isfocus (see below). I don't know how to send this re-INVITE request to continue to work on the flow...
F5 INVITE Bob -> Alice
INVITE sips:al...@client.atlanta.example.com SIP/2.0
Via: SIP/2.0/TLS client.biloxi.example.com:5061
;branch=z9hG4bKnashds
Max-Forwards: 70
From: Bob <sips:b...@biloxi.example.com>;tag=23431
To: Alice <sips:al...@atlanta.example.com>;tag=1234567
Call-ID: 1234...@atlanta.example.com
CSeq: 1024 INVITE
Contact: <sips:bob-...@client.biloxi.example.com>;isfocus
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Supported: replaces, gruu
Content-Length: ...
v=0
o=bob 2890844527 2890844528 IN IP4 client.biloxi.example.com
s=c=IN IP4 client.biloxi.example.com t=0 0 m=audio 49172 RTP/AVP 0 a=rtpmap:0 PCMU/8000
Thank you!
]. Bob then INVITEs Carol using the same Contact URI. Note that Bob could wait to re-INVITE Alice until after Carol has answered. Bob could also put Alice on hold before calling Carol. Message Details F1 INVITE Alice -> Bob INVITE sips...@biloxi.example.com SIP/2.0 Via: SIP/2.0/TLS client.atlanta.example.com:5061 ;branch=z9hG4bK74bf9Max-Forwards: 70 From: Alice <sips:...@atlanta.example.com>;tag=1234567 To: Bob <sips...@biloxi.example.com> Call-ID: 1234...@atlanta.example.com CSeq: 1 INVITE Contact: <sips:...@client.atlanta.example.com> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: ... v=0 o=alice 2890844526 2890844526 IN IP4 client.atlanta.example.com
s= c=IN IP4 client.atlanta.example.com t=0 0 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 F2 180 Ringing Bob -> Alice SIP/2.0 180 Ringing Via: SIP/2.0/TLS client.atlanta.example.com:5061
;branch=z9hG4bK74bf9 ;received=192.0.2.103 From: Alice <sips:...@atlanta.example.com>;tag=1234567 To: Bob <sips...@biloxi.example.com>;tag=23431 Call-ID: 1234...@atlanta.example.com CSeq: 1 INVITE Contact: <sips:b5...@biloxi.example.com> Content-Length: 0 F3 200 OK Bob -> Alice SIP/2.0 200 OK Via: SIP/2.0/TLS client.atlanta.example.com:5061 ;branch=z9hG4bK74bf9 ;received=192.0.2.103 From: Alice <sips:...@atlanta.example.com>;tag=1234567 To: Bob <sips...@biloxi.example.com>;tag=23431 Call-ID: 1234...@atlanta.example.com CSeq: 1 INVITE Contact: <sips:b5...@biloxi.example.com> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Supported: replaces, gruu Content-Type: application/sdp Content-Length: ... v=0 o=bob 2890844527 2890844527 IN IP4 client.biloxi.example.com s= c=IN IP4 client.biloxi.example.com
t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 F4 ACK Alice -> Bob ACK sips:b5...@biloxi.example.com SIP/2.0 Via: SIP/2.0/TLS client.atlanta.example.com:5061 ;branch=z9hG4bK74bfL Max-Forwards: 70 From: Alice <sips:...@atlanta.example.com>;tag=1234567 To: Bob <sips...@biloxi.example.com>;tag=23431 Call-ID: 1234...@atlanta.example.com CSeq: 1 ACK Content-Length: 0 /* Alice and Bob have established a session. Bob re-INVITEs, changing Contact URIs. */ F5 INVITE Bob -> Alice INVITE sips:...@client.atlanta.example.com SIP/2.0 Via: SIP/2.0/TLS client.biloxi.example.com:5061 ;branch=z9hG4bKnashds Max-Forwards: 70 From: Bob <sips...@biloxi.example.com>;tag=23431 To: Alice <sips:...@atlanta.example.com>;tag=1234567 Call-ID: 1234...@atlanta.example.com CSeq: 1024 INVITE Contact: <sips:bo...@client.biloxi.example.com>;isfocus Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Supported: replaces, gruu Content-Length: ... v=0 o=bob 2890844527 2890844528 IN IP4 client.biloxi.example.com s=
c=IN IP4 client.biloxi.example.com t=0 0 m=audio 49172 RTP/AVP 0 a=rtpmap:0 PCMU/8000 F6 200 OK Alice -> Bob SIP/2.0 200 OK Via: SIP/2.0/TLS client.biloxi.example.com:5061
;branch=z9hG4bKnashds7 ;received=192.0.2.113 From: Bob <sips...@biloxi.example.com>;tag=23431 To: Alice <sips:...@atlanta.example.com>;tag=1234567 Call-ID: 1234...@atlanta.example.com CSeq: 1024 INVITE Contact: <sips:...@client.atlanta.example.com> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: ... v=0 o=alice 2890844526 2890844526 IN IP4 client.atlanta.example.com s= c=IN IP4 client.atlanta.example.com t=0 0 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 F7 ACK Bob -> Alice ACK sips:...@client.atlanta.example.com SIP/2.0 Via: SIP/2.0/TLS client.biloxi.example.com:5061 ;branch=z9hG4bKnash3G Max-Forwards: 70 From: Bob <sips...@biloxi.example.com>;tag=23431 To: Alice <sips:...@atlanta.example.com>;tag=1234567 Call-ID: 1234...@atlanta.example.com
CSeq: 1024 ACK Content-Length: 0 /* Bob calls Carol. */
F8 INVITE Bob -> Carol INVITE sips:...@chicago.example.com SIP/2.0 Via: SIP/2.0/TLS client.biloxi.example.com:5061 ;branch=z9hG4bKnashJfd Max-Forwards: 70 From: Bob <sips...@biloxi.example.com>;tag=8675309 To: Carol <sips:...@chicago.example.com> Call-ID: sdjfd...@biloxi.example.com CSeq: 42 INVITE Contact: <sips:bo...@client.biloxi.example.com>;isfocus Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Supported: replaces, gruu Content-Type: application/sdp Content-Length: ... v=0 o=bob 28908445834 2890844834 IN IP4 client.biloxi.example.com
s= c=IN IP4 client.biloxi.example.com t=0 0 m=audio 48174 RTP/AVP 0 a=rtpmap:0 PCMU/8000 F9 180 Ringing Carol -> Bob SIP/2.0 200 OK Via: SIP/2.0/TLS client.biloxi.example.com:5061
;branch=z9hG4bKnashJfd ;received=192.0.2.113 From: Bob <sips...@biloxi.example.com>;tag=8675309 To: Carol <sips:...@chicago.example.com>;tag=341313 Call-ID: sdjfd...@biloxi.example.com CSeq: 42 INVITE Contact: <sips:...@client.chicago.example.com> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Supported: replaces Content-Length: 0 F10 200 OK Carol -> Bob SIP/2.0 200 OK Via: SIP/2.0/TLS client.biloxi.example.com:5061 ;branch=z9hG4bKnashJfd ;received=192.0.2.113 From: Bob <sips...@biloxi.example.com>;tag=8675309To: Carol <sips:...@chicago.example.com>;tag=341313 Call-ID: sdjfd...@biloxi.example.com CSeq: 42 INVITE Contact: <sips:...@client.chicago.example.com> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: ... v=0 o=carol 2890844922 2890844922 IN IP4 client.chicago.example.com s= c=IN IP4 client.chicago.example.com t=0 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 F11 ACK Bob -> Carol ACK sips:...@client.chicago.example.com SIP/2.0 Via: SIP/2.0/TLS client.biloxi.example.com:5061 ;branch=z9hG4bKnash431 Max-Forwards: 70 From: Bob <sips...@biloxi.example.com>;tag=8675309 To: Carol <sips:...@chicago.example.com>;tag=341313 Call-ID: sdjfd...@biloxi.example.com
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--José Luis Millán
José Luis Millán
Jun 8, 2019, 1:24:27 PM6/8/19
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What about this detail in the example?
This is, Bob sends to Alice Carol's audio and his own audio mixed, he sends to Calor Alice's audio and his own audio mixed. How do you plan to achieve it :-)?
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José Luis Millán
Leonardo Costa
Jun 10, 2019, 10:13:51 AM6/10/19
to JsSIP
Hello, José!
Doing a quick research, maybe we could achieve the mixing process using the Web Audio API (https://www.w3.org/TR/webaudio/)! I found this good example, Mix.js (http://kevvv.in/mix/), where they were able to mix 12 different tracks to play a music!
Taking this to our example (3-way conference), I think that we need to mix Bob's voice (microphone) with the audio that he receives from Alice, and finally send this mixed audio to Carol (and the reverse way too)! Bob will act like a bridge!
What do you think? Is it possible try to implement this feature on the JsSIP? I guess it would be a great plus to the library!
Thank you!
Em sábado, 8 de junho de 2019 14:24:27 UTC-3, José Luis Millán escreveu:
What about this detail in the example?"Bob is capable of media mixing in a 3-party call."This is, Bob sends to Alice Carol's audio and his own audio mixed, he sends to Calor Alice's audio and his own audio mixed. How do you plan to achieve it :-)?
El vie., 7 jun. 2019 a las 19:09, Leonardo Costa (<tradi...@gmail.com>) escribió:
Hello, José!In fact I want to follow the 3-way conference example flow as described on the RFC to make it work! I already have one conversation call (session) stablished between 2 users (Bob and Alice)! Following the RFC flow, the next step is to Bob send a re-INVITE to Alice (step F5), changing the Contatct URI and adding a the tag isfocus (see below). I don't know how to send this re-INVITE request to continue to work on the flow...
F5 INVITE Bob -> Alice
INVITE sips:...@client.atlanta.example.com SIP/2.0 Via: SIP/2.0/TLS client.biloxi.example.com:5061
Contact: <sips:bo...@client.biloxi.example.com>;isfocus
To view this discussion on the web visit https://groups.google.com/d/msgid/jssip/3ca9628b-f9b6-45ee-8fca-2f5bf1a7691b%40googlegroups.com.
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--José Luis Millán
Iñaki Baz Castillo
Jun 10, 2019, 10:17:19 AM6/10/19
to jssip
You can do it by mixing the audio tracks by yourself and then using the resulting media stream or its audio track with the PeerConnection.getSenders()[0].replaceTrack() API of WebRTC. Remember that the rtcSession.connection of JsSIP points to the underlying PeerConnection.
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José Luis Millán
Jun 10, 2019, 10:21:25 AM6/10/19
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El lun., 10 jun. 2019 a las 16:17, Iñaki Baz Castillo (<i...@aliax.net>) escribió:
You can do it by mixing the audio tracks by yourself and then using the resulting media stream or its audio track with the PeerConnection.getSenders()[0].replaceTrack() API of WebRTC. Remember that the rtcSession.connection of JsSIP points to the underlying PeerConnection.
Absolutetly agreed. Nothing to be done in the library.
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José Luis Millán
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Leonardo Costa
Jun 13, 2019, 9:25:35 AM6/13/19
to JsSIP
Hello, guys!
It worked for me! I was able to mix all received audio from the sessions with my own voice and the conference worked fine! I tested on Chrome and Firefox!
I'm just wondering why the RFC example says that we need to do all those steps before mix the audio (send the re-invite, change the contact header, add the isfocus tag, etc)...
Anyway, here a code example:
//function to create conference by mixing audio
//sessions => array with JsSIP.RTCSessions calls
//remoteAudioId => the ID of your <audio> element to play the received streams
function conference(sessions, remoteAudioId) {
//take all received tracks from the sessions you want to merge
var receivedTracks = [];
sessions.forEach(function(session) {
if(session !== null && session !== undefined) {
session.connection.getReceivers().forEach(function(receiver) {
receivedTracks.push(receiver.track);
});
}
});
//use the Web Audio API to mix the received tracks
var context = new AudioContext();
var allReceivedMediaStreams = new MediaStream();
sessions.forEach(function(session) {
if(session !== null && session !== undefined) {
var mixedOutput = context.createMediaStreamDestination();
session.connection.getReceivers().forEach(function(receiver) {
receivedTracks.forEach(function(track) {
allReceivedMediaStreams.addTrack(receiver.track);
var sourceStream = context.createMediaStreamSource(new MediaStream([track]));
sourceStream.connect(mixedOutput);
}
});
});
//mixing your voice with all the received audio
session.connection.getSenders().forEach(function(sender) {
var sourceStream = context.createMediaStreamSource(new MediaStream([sender.track]));
sourceStream.connect(mixedOutput);
});
session.connection.getSenders()[0].replaceTrack(mixedOutput.stream.getTracks()[0]);
}
});
//play all received stream to you
var remoteAudio = document.getElementById('remoteAudioId');
remoteAudio.srcObject = allReceivedMediaStreams;
var promiseRemote = remoteAudio.play();
if(promiseRemote !== undefined) {
promiseRemote.then(_ => {
console.log("playing all received streams to you");
}).catch(error => {
console.log(error);
});
}
}
Thanks!
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Iñaki Baz Castillo
Jun 13, 2019, 10:17:06 AM6/13/19
to jssip
Really cool :)
> To view this discussion on the web visit https://groups.google.com/d/msgid/jssip/b887869e-887c-474b-bf5b-4aa8b45c3ead%40googlegroups.com.
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Avi Charlop
Jun 29, 2020, 12:30:12 PM6/29/20
to JsSIP
Was really hoping to use this but it isn't working for me. Do you have any more details or code you you can share? Is there any peerconnection that needs to happen?
Viet Hung Le
Jun 30, 2020, 7:27:32 AM6/30/20
to JsSIP
The code Leonardo provided worked for me, but when bridge user agent press mute, then the others can't hear anything (also lost connection when he hang up), so I think use meshed conference (decentralized) will worked but it's so sophisticated and hard to manage connection.
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Bhupinder Gill
Jun 30, 2020, 1:05:39 PM6/30/20
to JsSIP
The code you provided worked for me , Thanks. I found some problem with it, Suppose user A accept call from user B and user C. Then user A put call in conference with B and C. I found that Audio from B to C and C to B does not work. B and C can not listen each other. Audio call from A to B , B to A, A to C and C to A work fine. Please see if you can help me here
Viet Hung Le
Jun 30, 2020, 11:22:55 PM6/30/20
to JsSIP
Maybe you did something wrong when you mix audio for senders. User B will listen mixed voices of A and C(from A) and C will listen voices of A and B (from A).
Bhupinder Gill
Jul 1, 2020, 1:54:19 AM7/1/20
to JsSIP
I found , that problem come when User A who started conference, put conference call on Mute. Only then user B and user C cannot listen each other. Is there other any way to put conference on mute or unmute?
To put conference on mute , I have applied below code.
if(sessions.length > 1) {
sessions.forEach((session) => {
session.mute();
});
}
if(sessions.length > 1) {
sessions.forEach((session) => {
session.unmute();
});
}
Viet Hung Le
Jul 1, 2020, 2:55:15 AM7/1/20
to JsSIP
As I told in the above post, because user A acts as bridge, when A mute, voice isn't sent, you have to use other topology: meshed or bridge (ref: https://www.cs.columbia.edu/sip/talks/sip-conferencing.pdf).
Hello, José!
s= c=IN IP4 <a href="http://client.chicago.example.com" rel="nofollow" target="_blank" onmousedown="this.href='http://www.google.com/url?q\x3dhttp%3A%2F%2Fclient.chicago.example.com\x26sa\x3dD\x26sntz\x3d1\x26usg\x3dAFQjCNHU_c1Z-KTSgTUMI9hZg1ZfXpoE3w';return true;" onclick="this.href='http://www.google.com/url?q\x3dhttp%3A%2F%2Fclient.chicago.example.com\x26sa\x3dD\x26sntz\x3d1\x26usg\x3dAFQjCNHU_c1Z-KTSgTUMI9hZg1ZfXpoE3w'
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Viet Hung Le
Jul 2, 2020, 7:41:55 AM7/2/20
to JsSIP
Hi Jose,
I want to implement decentralized conference call, when user A call B to join conference, can I use hold and REFER method to tell other user to make a call to B, or I have to send them message ?On Monday, July 2, 2018 at 9:43:26 PM UTC+7, Ashish Gautam wrote:
Hello Team,
How can we implement the 3 Ways Call Conference?
Message has been deleted
Iñaki Baz Castillo
Jul 2, 2020, 8:02:49 AM7/2/20
to jssip
On Thu, 2 Jul 2020 at 13:41, Viet Hung Le <leviet...@gmail.com> wrote:
Hi Jose,
This is a mailing list with many people subscribed to it. Please avoid direct mentions.
Vinícius Feres Belló
Aug 13, 2020, 9:47:20 AM8/13/20
to JsSIP
The code from Leonardo worked from me too. I'm using SIP.js + Asterisk. I just needed adjust session.connection to session.sessionDescriptionHandler.peerConnection. Thanks for sharing, it's really difficult find some examples about call conference.
Asad Abbas
Mar 22, 2021, 10:46:29 AM3/22/21
to JsSIP
Hi Bhupinder Gill,
I am getting the same issue when bridge user mutes the call than user A and B are unable to listen each other. Did you implement the mesh ? If yes can you give me the idea how I can do that?
Usman Akram
Jun 21, 2021, 4:58:55 PM6/21/21
to JsSIP
Hello Guys,
I'm facing the same issue of making conference call.
I tried above code but I think code is not completed.
Can somebody give me a clue how I can implement conference call.
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Jehanzaib Younis
Jun 30, 2022, 7:51:36 PM6/30/22
to JsSIP
Hi Usman,
Did you find a way to achieve this?
Roque
Aug 2, 2022, 10:35:22 PM8/2/22
to JsSIP
Hi Team, I'd like to make a conference between a security camera (It has a built-in softphone -> just video) and a doorbell (just audio) with a softphone (recieving audio and video, just sending audio and DTMF). Is it possible? What should I change/add on Leonardo's example?
Thanks in advance!
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