Hello all,
I am facing an issue with JsSIP not recognizing replies from Kamailio. the call sequence goes as follows:
INVITE -----------------------------> <-------------------------------SIP/2.0 100 Trying <-------------------------------SIP/2.0 180 Ringing <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------BYE 404 Not Found ---------------------->
When JsSIP receives ACK it prints an error: JsSIP:UA Request-URI does not point to us
From another thread with similar issue at https://groups.google.com/g/sip_js/c/uiaXS_qc2n8 it could be that JsSIP is not recognizing the GRUU is pointing towards it.
In the INVITE message I can see a line "Record-Route:
mailto:sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com mailto:sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com
<
mailto:sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com mailto:sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com
mailto:;expires=600;received=mailto:;gr=urn:uuid:b8a691ff-f678-4866-8cf0-780d670e7e33
<
mailto:sip:9747815015@erx-staging-q01.mydomain.com mailto:sip:Stg-CQD0r2-10020005@10.10.1.9
mailto:sip:Stg-CQD0r2-10020005@10.10.1.9 mailto:sip:9747815015@erx-staging-q01.mydomain.com
mailto:sip:Stg-CQD0r2-10020005@10.10.1.9 mailto:sip:9747815015@erx-staging-q01.mydomain.com
mailto:sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com
mailto:sip:Stg-CQD0r2-10020005@10.10.1.9 mailto:sip:9747815015@erx-staging-q01.mydomain.com
mailto:sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com
< mailto:sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com
mailto:sip:9747815015@erx-staging-q01.mydomain.com mailto:sip:Stg-CQD0r2-10020005@10.10.1.9
< mailto:sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com
mailto:sip:9747815015@erx-staging-q01.mydomain.com mailto:sip:Stg-CQD0r2-10020005@10.10.1.9
mailto:sip:Stg-CQD0r2-10020005@10.10.1.9 mailto:sip:9747815015@erx-staging-q01.mydomain.com
Hello,
is the REGISTER without a Contact URI or the message you pasted omitted it?
Cheers, Daniel
On 22.03.22 10:29, Xuo Guoto wrote:
Hello all,
I am facing an issue with JsSIP not recognizing replies from Kamailio. the call sequence goes as follows:
INVITE -----------------------------> <-------------------------------SIP/2.0 100 Trying <-------------------------------SIP/2.0 180 Ringing <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------SIP/2.0 200 OK ACK --------------------------------> <-------------------------------BYE 404 Not Found ---------------------->
When JsSIP receives ACK it prints an error: JsSIP:UA Request-URI does not point to us
From another thread with similar issue at https://groups.google.com/g/sip_js/c/uiaXS_qc2n8 it could be that JsSIP is not recognizing the GRUU is pointing towards it.
In the INVITE message I can see a line "Record-Route: " Is this normal?
I am at a loss to figure this out, and any help or hint to find out what could be wrong here would be very helpful.
The SIP messages are as follows, as seen from JsSIP:
Send -> REGISTER sip:erx-staging-q01.mydomain.com SIP/2.0 Via: SIP/2.0/WSS ol3dhprvu7jv.invalid;branch=z9hG4bK3674021 Max-Forwards: 69 To: sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com From: "User" sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com;tag=65u34oje2s Call-ID: b624vmbvuioma46354gmi5 CSeq: 1 REGISTER Contact: ;+sip.ice;reg-id=1;+sip.instance="";expires=600 Expires: 600 Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY Supported: path,gruu,outbound User-Agent: JsSIP 3.9.0 Content-Length: 0
Receive <- SIP/2.0 200 OK Via: SIP/2.0/WSS ol3dhprvu7jv.invalid;branch=z9hG4bK3674021;rport=23618;received=145.15.191.170 To: sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com;tag=ee7bee7ecdf2759680598685ea71a5eb.d5e40000 From: "User" sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com;tag=65u34oje2s Call-ID: b624vmbvuioma46354gmi5 CSeq: 1 REGISTER Contact: ;expires=600;received="sip:145.15.191.170:23618;transport=ws";pub-gruu="sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com mailto:;expires=600;received=;gr=urn:uuid:b8a691ff-f678-4866-8cf0-780d670e7e33";temp-gruu="sip:uloc-623966d2-1b2b74-79-435e33f9@erx-staging-q01.mydomain.com mailto:;gr=urn:uuid:b8a691ff-f678-4866-8cf0-780d670e7e33;gr";+sip.instance="";reg-id=1 Server: TLS Kamailio Server Content-Length: 0
Receive <- INVITE sip:93he4k0p@ol3dhprvu7jv.invalid;transport=ws SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/WSS 68.19.159.72:443;branch=z9hG4bKc3f.2a0c8eb470c9427d6def7b8eaa3e3f8b.0 Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.y7M.y7MmZfMxvwMxyAzweI36KYpEKqH.FAOBFAOBF7M.yXKFcQgSI6zweAowe4H.NAMxyX3heroEWvH9vsCFN43q1PCEKIMxWr3BN6O.gqMlq1WxerMJ4ZW6aqO.MszSV6z.PwMlqAgc** From: "User2" sip:9747815015@erx-staging-q01.mydomain.com;tag=27ec81cf-9ccc-45e6-b712-fe23337ed7d9 To: sip:Stg-CQD0r2-10020005@10.10.1.9 Contact: Call-ID: 518500d1-9813-4af8-a249-981ca2ee8a4b CSeq: 19353 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub, histinfo Session-Expires: 1800 Min-SE: 90 Max-Forwards: 69 User-Agent: Asterisk PBX 18.8.0 Content-Type: application/sdp Content-Length: 881
Send -> SIP/2.0 100 Trying Via: SIP/2.0/WSS 68.19.159.72:443;branch=z9hG4bKc3f.2a0c8eb470c9427d6def7b8eaa3e3f8b.0 Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.y7M.y7MmZfMxvwMxyAzweI36KYpEKqH.FAOBFAOBF7M.yXKFcQgSI6zweAowe4H.NAMxyX3heroEWvH9vsCFN43q1PCEKIMxWr3BN6O.gqMlq1WxerMJ4ZW6aqO.MszSV6z.PwMlqAgc** To: sip:Stg-CQD0r2-10020005@10.10.1.9 From: "User2" sip:9747815015@erx-staging-q01.mydomain.com;tag=27ec81cf-9ccc-45e6-b712-fe23337ed7d9 Call-ID: 518500d1-9813-4af8-a249-981ca2ee8a4b CSeq: 19353 INVITE Supported: timer,gruu,ice,replaces,outbound Content-Length: 0
Send -> SIP/2.0 180 Ringing Record-Route: Record-Route: Via: SIP/2.0/WSS 68.19.159.72:443;branch=z9hG4bKc3f.2a0c8eb470c9427d6def7b8eaa3e3f8b.0 Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.y7M.y7MmZfMxvwMxyAzweI36KYpEKqH.FAOBFAOBF7M.yXKFcQgSI6zweAowe4H.NAMxyX3heroEWvH9vsCFN43q1PCEKIMxWr3BN6O.gqMlq1WxerMJ4ZW6aqO.MszSV6z.PwMlqAgc** To: sip:Stg-CQD0r2-10020005@10.10.1.9;tag=6gc6gshfkb From: "User2" sip:9747815015@erx-staging-q01.mydomain.com;tag=27ec81cf-9ccc-45e6-b712-fe23337ed7d9 Call-ID: 518500d1-9813-4af8-a249-981ca2ee8a4b CSeq: 19353 INVITE Contact: sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com;gr=urn:uuid:b8a691ff-f678-4866-8cf0-780d670e7e33 Supported: timer,gruu,ice,replaces,outbound Content-Length: 0
Send -> SIP/2.0 200 OK Record-Route: Record-Route: Via: SIP/2.0/WSS 68.19.159.72:443;branch=z9hG4bKc3f.2a0c8eb470c9427d6def7b8eaa3e3f8b.0 Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.y7M.y7MmZfMxvwMxyAzweI36KYpEKqH.FAOBFAOBF7M.yXKFcQgSI6zweAowe4H.NAMxyX3heroEWvH9vsCFN43q1PCEKIMxWr3BN6O.gqMlq1WxerMJ4ZW6aqO.MszSV6z.PwMlqAgc** To: sip:Stg-CQD0r2-10020005@10.10.1.9;tag=6gc6gshfkb From: "User2" sip:9747815015@erx-staging-q01.mydomain.com;tag=27ec81cf-9ccc-45e6-b712-fe23337ed7d9 Call-ID: 518500d1-9813-4af8-a249-981ca2ee8a4b CSeq: 19353 INVITE Contact: sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com;gr=urn:uuid:b8a691ff-f678-4866-8cf0-780d670e7e33 Session-Expires: 1800;refresher=uas Supported: timer,gruu,ice,replaces,outbound Content-Type: application/sdp Content-Length: 947
Receive <- ACK sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com;gr=urn:uuid:b8a691ff-f678-4866-8cf0-780d670e7e33 SIP/2.0 Via: SIP/2.0/WSS 68.19.159.72:443;branch=z9hG4bKc3f.244f565f3d688006fb9c33138458f554.0 Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.y7M.y7MmZfMxvwMxyAzweI36KYpEKqH.FAOBFAOBF7M.yXKFcQgSW6zweAowe4H.NAMxyX3heroEWvH9vsCFN43q1PCEjuMlWEWB0rO.aJgBc1WSPAMG4Z3RjLO.pqWSPlMRFwWEergc** From: "User2" sip:9747815015@erx-staging-q01.mydomain.com;tag=27ec81cf-9ccc-45e6-b712-fe23337ed7d9 To: sip:Stg-CQD0r2-10020005@10.10.1.9;tag=6gc6gshfkb Call-ID: 518500d1-9813-4af8-a249-981ca2ee8a4b CSeq: 19353 ACK Max-Forwards: 69 User-Agent: Asterisk PBX 18.8.0 Content-Length: 0
tryit-jssip.js:8 JsSIP:UA Request-URI does not point to us +40s
Receive <- BYE sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com;gr=urn:uuid:b8a691ff-f678-4866-8cf0-780d670e7e33 SIP/2.0 Via: SIP/2.0/WSS 68.19.159.72:443;branch=z9hG4bK93f.8fbee7f4747c4d72cc73b403fe51c081.0 Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.y7M.y7MmZfMxvwMxyAzweI36KYpEKqH.FAOBFAOBF7M.yXKFcQgSW6zweAowe4H.NAMxyX3heroEWvH9vsCFN43q1PCBKq36PfMBysO.cA3631Wx3fMmuJ36FuO.PuWEFsgxc4MSKB3A** From: "User2" sip:9747815015@erx-staging-q01.mydomain.com;tag=27ec81cf-9ccc-45e6-b712-fe23337ed7d9 To: sip:Stg-CQD0r2-10020005@10.10.1.9;tag=6gc6gshfkb Call-ID: 518500d1-9813-4af8-a249-981ca2ee8a4b CSeq: 19354 BYE Reason: Q.850;cause=16 Max-Forwards: 69 User-Agent: Asterisk PBX 18.8.0 Content-Length: 0
tryit-jssip.js:8 JsSIP:UA Request-URI does not point to us +3s
Send -> SIP/2.0 404 Not Found Via: SIP/2.0/WSS 68.19.159.72:443;branch=z9hG4bK93f.8fbee7f4747c4d72cc73b403fe51c081.0 Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.y7M.y7MmZfMxvwMxyAzweI36KYpEKqH.FAOBFAOBF7M.yXKFcQgSW6zweAowe4H.NAMxyX3heroEWvH9vsCFN43q1PCBKq36PfMBysO.cA3631Wx3fMmuJ36FuO.PuWEFsgxc4MSKB3A** To: sip:Stg-CQD0r2-10020005@10.10.1.9;tag=6gc6gshfkb From: "User2" sip:9747815015@erx-staging-q01.mydomain.com;tag=27ec81cf-9ccc-45e6-b712-fe23337ed7d9 Call-ID: 518500d1-9813-4af8-a249-981ca2ee8a4b CSeq: 19354 BYE Content-Length: 0
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Hi,
It seems when I paste the message in the web client, it got removed. Now trying again in text mode.
REGISTER sip:erx-staging-q01.mydomain.com SIP/2.0 Via: SIP/2.0/WSS ol3dhprvu7jv.invalid;branch=z9hG4bK3674021 Max-Forwards: 69 To: sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com From: "User" sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com;tag=65u34oje2s Call-ID: b624vmbvuioma46354gmi5 CSeq: 1 REGISTER Contact: sip:93he4k0p@ol3dhprvu7jv.invalid;transport=ws;+sip.ice;reg-id=1;+sip.instance="urn:uuid:b8a691ff-f678-4866-8cf0-780d670e7e33";expires=600 Expires: 600 Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY Supported: path,gruu,outbound User-Agent: JsSIP 3.9.0 Content-Length: 0
I hadn't noticed that some text was removed by the client.
X.
------- Original Message -------
On Thursday, March 24th, 2022 at 6:31 PM, Daniel-Constantin Mierla miconda@gmail.com wrote:
Hello,
is the REGISTER without a Contact URI or the message you pasted omitted it?
Cheers,
Daniel
On 22.03.22 10:29, Xuo Guoto wrote:
Hello all,
I am facing an issue with JsSIP not recognizing replies from Kamailio. the call sequence goes as follows:
INVITE -----------------------------><-------------------------------SIP/2.0 100 Trying<-------------------------------SIP/2.0 180 Ringing<-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------BYE404 Not Found ---------------------->
When JsSIP receives ACK it prints an error: JsSIP:UA Request-URI does not point to us
This is the ACK packet that is not getting recognized by JsSIP
ACK sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com;gr=urn:uuid:b8a691ff-f678-4866-8cf0-780d670e7e33 SIP/2.0 Via: SIP/2.0/WSS 68.19.59.72:443;branch=z9hG4bKc3f.244f565f3d688006fb9c33138458f554.0 Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.y7M.y7MmZfMxvwMxyAzweI36KYpEKqH.FAOBFAOBF7M.yXKFcQgSW6zweAowe4H.NAMxyX3heroEWvH9vsCFN43q1PCEjuMlWEWB0rO.aJgBc1WSPAMG4Z3RjLO.pqWSPlMRFwWEergc** From: "User2" sip:9747815015@erx-staging-q01.mydomain.com;tag=27ec81cf-9ccc-45e6-b712-fe23337ed7d9 To: sip:Stg-CQD0r2-10020005@10.10.1.9;tag=6gc6gshfkb Call-ID: 518500d1-9813-4af8-a249-981ca2ee8a4b CSeq: 19353 ACK Max-Forwards: 69 User-Agent: Asterisk PBX 18.8.0 Content-Length: 0
tryit-jssip.js:8 JsSIP:UA Request-URI does not point to us +40s
------- Original Message -------
On Thursday, March 24th, 2022 at 8:20 PM, Xuo Guoto xuoguoto@protonmail.com wrote:
Hi,
It seems when I paste the message in the web client, it got removed. Now trying again in text mode.
REGISTER sip:erx-staging-q01.mydomain.com SIP/2.0
Via: SIP/2.0/WSS ol3dhprvu7jv.invalid;branch=z9hG4bK3674021
Max-Forwards: 69
To: sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com
From: "User" sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com;tag=65u34oje2s
Call-ID: b624vmbvuioma46354gmi5
CSeq: 1 REGISTER
Contact: sip:93he4k0p@ol3dhprvu7jv.invalid;transport=ws;+sip.ice;reg-id=1;+sip.instance="urn:uuid:b8a691ff-f678-4866-8cf0-780d670e7e33";expires=600
Expires: 600
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: path,gruu,outbound
User-Agent: JsSIP 3.9.0
Content-Length: 0
I hadn't noticed that some text was removed by the client.
X.
------- Original Message -------
On Thursday, March 24th, 2022 at 6:31 PM, Daniel-Constantin Mierla miconda@gmail.com wrote:
Hello,
is the REGISTER without a Contact URI or the message you pasted omitted it?
Cheers,
Daniel
On 22.03.22 10:29, Xuo Guoto wrote:
Hello all,
I am facing an issue with JsSIP not recognizing replies from Kamailio. the call sequence goes as follows:
INVITE -----------------------------><-------------------------------SIP/2.0 100 Trying<-------------------------------SIP/2.0 180 Ringing<-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------BYE404 Not Found ---------------------->
When JsSIP receives ACK it prints an error: JsSIP:UA Request-URI does not point to us
Did some more digging into the source code of JsSIP. The condition that triggers the error message is: https://github.com/versatica/JsSIP/blob/3ab1fa7c8e09231c41ca21657bf962323906... /** * Request reception */ receiveRequest(request) { const method = request.method;
// Check that request URI points to us. if (request.ruri.user !== this._configuration.uri.user && request.ruri.user !== this._contact.uri.user) { logger.debug('Request-URI does not point to us'); if (request.method !== JsSIP_C.ACK) { request.reply_sl(404); }
return; }
If I understand correctly if ruri.user is neither user in config or user in contact the request gets rejected.
R-URI is: ACK sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com;gr=urn:uuid:b8a691ff-f678-4866-8cf0-780d670e7e33 SIP/2.0
The SIP URI configured in tryjsip is sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com
This does not match, but if I change the SIP URI to sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com (all small case) it works fine. I am not sure if this is a bug or violates the any RFC
Sent with ProtonMail secure email.
------- Original Message -------
On Thursday, March 24th, 2022 at 9:10 PM, Xuo Guoto xuoguoto@protonmail.com wrote:
This is the ACK packet that is not getting recognized by JsSIP
ACK sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com;gr=urn:uuid:b8a691ff-f678-4866-8cf0-780d670e7e33 SIP/2.0
Via: SIP/2.0/WSS 68.19.59.72:443;branch=z9hG4bKc3f.244f565f3d688006fb9c33138458f554.0
Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.y7M.y7MmZfMxvwMxyAzweI36KYpEKqH.FAOBFAOBF7M.yXKFcQgSW6zweAowe4H.NAMxyX3heroEWvH9vsCFN43q1PCEjuMlWEWB0rO.aJgBc1WSPAMG4Z3RjLO.pqWSPlMRFwWEergc**
From: "User2" sip:9747815015@erx-staging-q01.mydomain.com;tag=27ec81cf-9ccc-45e6-b712-fe23337ed7d9
To: sip:Stg-CQD0r2-10020005@10.10.1.9;tag=6gc6gshfkb
Call-ID: 518500d1-9813-4af8-a249-981ca2ee8a4b
CSeq: 19353 ACK
Max-Forwards: 69
User-Agent: Asterisk PBX 18.8.0
Content-Length: 0
tryit-jssip.js:8 JsSIP:UA Request-URI does not point to us +40s
------- Original Message -------
On Thursday, March 24th, 2022 at 8:20 PM, Xuo Guoto xuoguoto@protonmail.com wrote:
Hi,
It seems when I paste the message in the web client, it got removed. Now trying again in text mode.
REGISTER sip:erx-staging-q01.mydomain.com SIP/2.0
Via: SIP/2.0/WSS ol3dhprvu7jv.invalid;branch=z9hG4bK3674021
Max-Forwards: 69
To: sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com
From: "User" sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com;tag=65u34oje2s
Call-ID: b624vmbvuioma46354gmi5
CSeq: 1 REGISTER
Contact: sip:93he4k0p@ol3dhprvu7jv.invalid;transport=ws;+sip.ice;reg-id=1;+sip.instance="urn:uuid:b8a691ff-f678-4866-8cf0-780d670e7e33";expires=600
Expires: 600
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: path,gruu,outbound
User-Agent: JsSIP 3.9.0
Content-Length: 0
I hadn't noticed that some text was removed by the client.
X.
------- Original Message -------
On Thursday, March 24th, 2022 at 6:31 PM, Daniel-Constantin Mierla miconda@gmail.com wrote:
Hello,
is the REGISTER without a Contact URI or the message you pasted omitted it?
Cheers,
Daniel
On 22.03.22 10:29, Xuo Guoto wrote:
Hello all,
I am facing an issue with JsSIP not recognizing replies from Kamailio. the call sequence goes as follows:
INVITE -----------------------------><-------------------------------SIP/2.0 100 Trying<-------------------------------SIP/2.0 180 Ringing<-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------SIP/2.0 200 OKACK --------------------------------><-------------------------------BYE404 Not Found ---------------------->
When JsSIP receives ACK it prints an error: JsSIP:UA Request-URI does not point to us
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Daniel-Constantin Mierla -
Xuo Guoto