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| 08:43.52 | skrusty | morning |
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| 09:01.25 | oej | Morning folks! |
| 09:01.39 | skrusty | morning |
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| 14:03.59 | hurdman | hi, i read the src code of chan_sip.c , i had debug into, but i can't understand why i can see asterisk receive my progress message but never re-send it to the other client. |
| 14:04.20 | hurdman | into handle_response_invite |
| 14:07.53 | hurdman | http://sebsauvage.net/paste/?e09b10d4b871b8cb#IrlmLYZzWCoWRQncbNUhsZhAgz8nJkFzI9dEGwD4N0Q= |
| 14:08.36 | hurdman | where is the error ? |
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| 14:09.38 | oej | There's a setting for progressinband that you can play around with. |
| 14:09.46 | oej | Normally it should be forwarded over the bridge. |
| 14:10.35 | hurdman | i wan't to have the progress forwarded by asterisk |
| 14:11.06 | jsmith | hurdman: I can think of several scenarios where that might happen -- for example, Alice calls Asterisk, Asterisk sends Alice a 100 Trying, Asterisk calls Bob, Bob sends back a 100 Trying... |
| 14:11.08 | oej | That's the default behaviour. |
| 14:11.22 | jsmith | hurdman: Asterisk won't send Alice another 100 Trying, because it's already sent one... |
| 14:11.37 | jsmith | hurdman: But without more information about what you're trying to do, I'm at a loss on how to further help you |
| 14:11.45 | oej | 100 Trying is ALWAYS hop by hop and NEVER forwarded jsmith. |
| 14:11.57 | hurdman | it's not what i meen, Alice send a progress to asterisk , asterisk receive the progress, but never send it to bob |
| 14:12.00 | oej | (Taking a note in my SIP masterclass participant directory. Jsmith --) |
| 14:12.16 | jsmith | oej: Well, not forwarded -- but sending a new one in response to an incoming one -- sorry for not being precise :-p |
| 14:12.19 | oej | The default behaviour is to forward it. |
| 14:12.35 | oej | The progress message |
| 14:13.01 | hurdman | i can see the progress from alice to asterisk (tpcdump ) and into src code of chan_sip but never forwarded |
| 14:13.01 | jsmith | hurdman: Can you give us more details? Who is calling who, which progress message in particular you're looking at (100? 180? 183?), etc.? |
| 14:13.21 | hurdman | have you read my pastebin ? |
| 14:13.46 | oej | 180 is not progress, 183 is (jsmith --) |
| 14:13.51 | oej | he he |
| 14:13.52 | hurdman | :) |
| 14:14.05 | jsmith | hurdman: Yes, but it's hard to follow, because I don't have the line numbers for the source code to match up with your ast_log statements |
| 14:14.08 | oej | An incoming 183 is forwarded to the bridge and handled as "early media" |
| 14:14.24 | jsmith | oej: Wow, I'm being really imprecise today -- thanks for calling me out on it -- I mean "provisional responses", not "progress" |
| 14:14.26 | hurdman | jsmith: i have paste the src code before my log ;) |
| 14:14.29 | oej | The call is not in up state but there's a bridge kind-of |
| 14:14.34 | hurdman | it's about 183, |
| 14:14.47 | jsmith | hurdman: OK, so let me make sure I understand you. |
| 14:15.21 | jsmith | hurdman: Alice calls Asterisk... Do you see any provisional responses (100,180,183) from Asterisk to Alice *before* Asterisk attempts to call Bob? |
| 14:16.06 | hurdman | i can't see any 183 from asterisk to Alice |
| 14:16.21 | jsmith | What *do* you see? |
| 14:16.26 | jsmith | Do you see a 100 Trying? |
| 14:16.32 | jsmith | Do you see a 180 Ringing? |
| 14:16.49 | hurdman | yes |
| 14:16.55 | jsmith | Again, I'm just trying to make sure I understand your situtation |
| 14:16.56 | hurdman | both 100 and 180 |
| 14:17.03 | jsmith | OK, so let's dig a bit further |
| 14:17.07 | hurdman | and i can established the call with a well forwarded rtp |
| 14:17.36 | jsmith | Alice calls Asterisk, Asterisk sends Alice a 100 Trying... Asterisk then attempts to call Bob |
| 14:17.45 | jsmith | Two questions: |
| 14:17.45 | hurdman | yes, it is |
| 14:18.13 | jsmith | 1) What does the Dial() statement in the dialplan look like when Asterisk attempts to dial Bob? (For example, is the 'r' option used?) |
| 14:18.26 | jsmith | 2) What provisional responses (100, 180, 183) does Bob send back to Asterisk? |
| 14:19.14 | hurdman | 1) i have try to add the r options, but nothing different appends |
| 14:19.45 | jsmith | (Please don't use the 'r' option -- every time you use the 'r' option, a kitten on the internet gets killed) |
| 14:20.37 | hurdman | 2) Bob send 100,180 and when i push a button to request an early media, it send a 183 that asterisk receive (the one i can see into my debug log =) |
| 14:21.06 | hurdman | my problem is that asterisk never relays that 183 to my dear alice |
| 14:22.36 | oej | Run "sip show peer" on alice and see the setting for progress |
| 14:23.15 | hurdman | you meen sip show settings ? |
| 14:23.49 | oej | "sip show peer alice" |
| 14:24.00 | oej | sip show settings is always nice too |
| 14:24.19 | hurdman | http://sebsauvage.net/paste/?e07b5425636f7a72#Kdj3LKJTRPFwMXybMKSCnbDy3SWvNXg/+MSwJPmFBtM= |
| 14:25.30 | hurdman | http://sebsauvage.net/paste/?8e5d7026fd022cd4#CXIaddlagwYBNy+Ze6vN6PzU5FdRFBggxxv3RciErZ0= |
| 14:25.40 | oej | I am pretty sure there is a setting for progressinband that controls 183 but I don't see it. Have no time to check right now, sorry. |
| 14:26.37 | hurdman | if you meen prematuremedia=no and progressinband=yes , i have tested all possible value |
| 14:27.03 | oej | Which version of Asterisk |
| 14:27.19 | hurdman | 11-5-1 |
| 14:28.02 | oej | Haven't got much experience of that version |
| 14:28.27 | oej | But in older versions early media worked all right. I have worked a bit with that as it is a requirement in telco networks. |
| 14:28.50 | hurdman | i have made a diff of chan_sip between 11-5 and 13, but nothing looks change for progress |
| 14:29.21 | hurdman | i ever had early sound from asterisk to alice during a call with real T2 |
| 14:29.37 | hurdman | but i have never seen early media works during a SIP call |
| 14:29.46 | oej | Test with 1.8 and see if there's a change |
| 14:30.20 | hurdman | ever tested, it's why because i have upgrade to 11-5, i was on 1.8 :/ |
| 14:33.40 | hurdman | into chan_sip.c, where does asterisk effectively send a progress to someone ? |
| 14:35.55 | hurdman | transmit_response_with_t38_sdp <= here ? |
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| 14:37.47 | jsmith | This isn't t38, right? |
| 14:39.37 | hurdman | transmit_response_with_sdp <= yes the other one ^^" |
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| 14:41.46 | jsmith | hurdman: I don't understand what you mean when you say "when I push a button to request an early media" |
| 14:41.55 | jsmith | hurdman: Can you give me more details on what you're trying to do there? |
| 14:42.35 | jsmith | is in a conference call for work, but will keep trying to help hurdman as he has time |
| 14:42.49 | hurdman | jsmith: thx o/ |
| 14:43.28 | hurdman | jsmith: i have a sort of linphone that send a progress (183) when i push a button |
| 14:44.18 | jsmith | Ok, and in your scenario, this softphone is Bob? |
| 14:44.48 | hurdman | both bob and alice, asterisk is in the middle |
| 14:45.15 | jsmith | OK, so let me make sure I understand |
| 14:45.55 | jsmith | Alice calls Asterisk, Asterisk calls Bob -- before Bob answer the call, he pushes a button to send early media back to Asterisk (and hence back to Alice) |
| 14:45.58 | jsmith | Does that sound right? |
| 14:47.28 | hurdman | yes ! |
| 14:47.55 | jsmith | Do you have a packet trace you can share with me? |
| 14:48.03 | hurdman | yes of course |
| 14:48.13 | jsmith | I'll send you my email in a private message |
| 14:48.21 | hurdman | ok, thx |
| 14:51.34 | jsmith | I've gotta prep for a meeting here in a few minutes, but I'll see if I can setup a test in my lab a bit later |
| 14:55.14 | hurdman | jsmith: send ! (thx), no problem, i will be around here for a while. |
| 14:57.16 | jsmith | hurdman: You don't have that in pcap format by chance, do you? |
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| 14:58.10 | hurdman | i can do that :) |
| 14:58.44 | oej | Jsmith: To debug this you need to see what's going on in asterisk. pcap will just confirm that the 183 is missing, not why. |
| 15:03.15 | hurdman | jsmith: done ! |
| 15:04.45 | jsmith | oej: I understand that :-) I'm just trying to reproduce the same call flow in my lab first, then I'll dive into more details |
| 15:06.14 | oej | great |
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| 17:46.12 | file | falls over |
| 17:48.41 | mjordan | weebles wobble |
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| 17:54.51 | hurdman | sorry i have had a split |
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| 18:47.09 | jsmith | hurdman: I just sent you an email with the results of my testing -- unfortunately, the test worked just fine in my lab :-/ |
| 18:50.18 | hurdman | ouch |
| 18:50.36 | hurdman | i take a look and back tomorow ;) thx for your help ! |
| 18:50.49 | jsmith | No problem |
| 18:51.01 | jsmith | At a minimum, it should give you hope that we can get this working for you |
| 18:51.09 | hurdman | i haven't send the Ringing during the call, perhaps it's that |
| 18:51.11 | jsmith | (and see how I emulated your linphone with a bit of Asterisk dialplan) |
| 18:51.20 | hurdman | can' you send your sip.conf please ? |
| 18:51.34 | jsmith | Sure... on Bob? |
| 18:51.43 | jsmith | or on Asterisk, or both? |
| 18:51.50 | jsmith | They're both very simple |
| 18:51.53 | hurdman | asterisk or both :) |
| 18:52.15 | jsmith | Here's the one from Bob: |
| 18:52.18 | jsmith | [connection] |
| 18:52.18 | jsmith | type=peer |
| 18:52.18 | jsmith | host=192.168.55.100 |
| 18:52.18 | jsmith | insecure=port,invite |
| 18:52.18 | jsmith | context=earlymedia |
| 18:52.20 | jsmith | <PROTECTED> |
| 18:52.34 | jsmith | And here's the one from Asterisk (the connection to Bob) |
| 18:53.09 | jsmith | [Bob] |
| 18:53.09 | jsmith | type=peer |
| 18:53.09 | jsmith | host=192.168.55.118 |
| 18:53.09 | jsmith | defaultuser=connection |
| 18:58.13 | hurdman | ok thx |
| 19:03.06 | mjordan | kharwell: file: ping |
| 19:03.09 | mjordan | specifically on https://reviewboard.asterisk.org/r/4216 |
| 19:03.18 | file | oui |
| 19:03.34 | mjordan | while I don't disagree with the naming change of remote to incoming, it feels like an unnecessary change for the vast majority of channel drivers |
| 19:03.53 | mjordan | not to mention changing an API in existing branches for somewhat pedantic reasons (outside of the SIP/PJSIP stuff we're seeing) |
| 19:04.19 | file | ack |
| 19:04.35 | file | if internally it is changed and the API changes are merely additions then I'm down with that |
| 19:05.15 | mjordan | How about if we leave the existing virtual table function alone as well as the existing API calls, but keep ast_rtp_instance_set_requested_source_address (and its variants)? |
| 19:05.36 | mjordan | the only thing with the virt table is *if* someone had their own RTP implementation, we'd be breaking them |
| 19:05.56 | file | requested source address? |
| 19:06.08 | mjordan | I believe those are the new RTP engine calls |
| 19:06.25 | file | requested target address |
| 19:06.36 | mjordan | ah, yes |
| 19:06.43 | mjordan | I made a mash up there ;-) |
| 19:06.56 | file | indeed - and like much music... not a good one |
| 19:07.01 | mjordan | that will also make that review a lot smaller :-) |
| 19:07.18 | kharwell | mjordan: okay I'll remove the changes to the api(s) and just the new function |
| 19:07.44 | kharwell | *add the new function that is |
| 19:07.58 | mjordan | kharwell: sorry for the back & forth - but until I saw the review, I didn't realize how much of that was just a name change |
| 19:08.22 | mjordan | I do think the documentation of the APIs should be "enhanced" to very clearly spell out what is a "remote" and what is a "requested target" address |
| 19:08.31 | file | yeah my fault - trying to bring some sanity for the next hopeless person to be thrown at it |
| 20:26.13 | skrusty | evening |
| 20:35.03 | file | g'evening |
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